Hi All...
I would like to interface 4 BRI lines to Asterisk. I looked at Digium's
hardware list and, although they have solutions for PRI and T1, I didn't
see anything for BRI. I would like to avoid ISDN4Linux if possible. Does
anyone know of any hardware suppoted by Asterisk I can use for
No, this is really a bug. Actually, a nasty, untraceable bug.
Check bugnotes for details.
Michael.
Brian West wrote:
Was this bug fixed or was it really a bug. I'm reading the bug notes and
it doesn't appear to be a bug in asterisk from what Mark said on the
notes.
bkw
On Wed, 11 Feb 2004,
Look at this:
http://www.junghanns.net/asterisk/page17.html
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Jim Archer
Gesendet: Montag, 16. Februar 2004 10:11
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] Need to interface to
Hi Jim,
we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN.
You can find more information about it at:
http://www.junghanns.net/asterisk/page17.html
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30
I forgot to mention, I am in North America.
--On Monday, February 16, 2004 4:10 AM -0500 Jim Archer [EMAIL PROTECTED]
wrote:
Hi All...
I would like to interface 4 BRI lines to Asterisk. I looked at Digium's
hardware list and, although they have solutions for PRI and T1, I didn't
see anything
Did you try the latest version (0.5.9)?
Michael.
Tomica Crnek wrote:
Hi everyone,
Does anyone know the answer for this situation? I have Asterisk with E1
PRI links, with SIP phones registered to Asterisk and with h.323
connection to Cisco CallManager. I am using oh323. I think I have a
Does anyone have any ideas on how to stop these messages from the SJPhone?
everything i've seen says they're harmless, but they're filling my console
and if anyone has any ides on how to make them go away i would be
appreciative.
thanks,
yair
Jim Archer ([EMAIL PROTECTED]) wrote:
Hi All...
I would like to interface 4 BRI lines to Asterisk. I looked at Digium's
hardware list and, although they have solutions for PRI and T1, I didn't
see anything for BRI. I would like to avoid ISDN4Linux if possible. Does
anyone know of any
Hi,
-Original Message-
I would like to interface 4 BRI lines to Asterisk. I looked
at Digium's
hardware list and, although they have solutions for PRI and
T1, I didn't
see anything for BRI. I would like to avoid ISDN4Linux if
possible. Does
anyone know of any hardware
Hi,
On Mon, 16 Feb 2004 at 04:10, Jim Archer wrote:
I would like to interface 4 BRI lines to Asterisk. I looked at
Digium's hardware list and, although they have solutions for PRI and
T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux
if possible. Does anyone know of any
Hi,
I'm using SIP channel, and i would like to authenticate users with a LDAP
server. Is this feature implemented in Asterik? I have read some posts about
it, but i don't know if it's currently available.
Thank you very much.
.G
Hi Jim,
i forgot to mention that the drivers do not yet support NI-1, but will
support it in the near future. Until then the only solution for you
will be the Eicon Diva Server 4BRI-8M and chan_capi.
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 -
so you need North America ISDN, not EuroIsdn.
The only way is a diva server card with capi driver,
Klaus zapBri doesn't support NI (as far as I know)
Matteo.
Il lun, 2004-02-16 alle 10:53, Jim Archer ha scritto:
I forgot to mention, I am in North America.
--On Monday, February 16, 2004 4:10
This has been talked about many times.
They are harmless as far as anyone know.
To get rid of them, get a different client. Or tell SJPhone to change
there code.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yair hakak
Sent: Monday, February 16, 2004
After finding a spot to put the Zhone Zplex so that the fan noise
doesn't annoy anyone, I've got everything working to an acceptable level
except for call transfers. No matter what I do the Zhone doesn't seem to
be passing 'flash' key presses on to asterisk, ie whenever I try to
transfer a call,
Klaus-Peter Junghanns [EMAIL PROTECTED] said:
we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN.
One thing I'd like to know about this card: Echo Cancellation? I've
replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is
remarkable...
(OP: there's also a 4BRI from Eicon,
I have just installed 0.5.9. Up to this moment it didn't crash :P
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Manousos
Sent: Monday, February 16, 2004 11:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk - oh323 - Cisco
Does it work? I
can't find if there is a config option in pppd config to make pppd authenticate
user against radius server.
Tomica
ah ah what is the trasnfer hook time for
you phone? Usa phones seems to have a very long
time... like 1secs, and that's the default
for zaptel... here in italy we have flash time
about 120 msec, instead
so, try that:
edit zaptel.h in your zaptel src dir,
search for ZT_DEFAULT_RXFLASHTIME
Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto:
Klaus-Peter Junghanns [EMAIL PROTECTED] said:
we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN.
One thing I'd like to know about this card: Echo Cancellation? I've
replaced by Fritz!Card PCI by a Diva Server 2M, and the
Kent Williams wrote:
After finding a spot to put the Zhone Zplex so that the fan noise
doesn't annoy anyone, I've got everything working to an acceptable level
except for call transfers. No matter what I do the Zhone doesn't seem to
be passing 'flash' key presses on to asterisk, ie whenever I try
Miguel,
IPC5000 doesn't support G729 (8 kbps) (it only support G711 64kbps)
Be carefull with what you buy.
Hector.
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Sunday, February 15, 2004 9:38 AM
Subject: [Asterisk-Users] Wifi
Well, I've made a little progress with this.
In zaptel.conf I set em=11 (11 is the channel).
In zapata.conf I set signalling=em_w and usecallerid=no
In extensions.conf I set 311,1,Dial(Zap/11,15)
In the Adtran, I set the port to FXS DPO.
At the beginning of each page, I hear two DTMF 1 tones (the
Hello,
I have configured 8500 to access
voicemailmain. With whatpriority does the control exit when the user
hangsup the phone without pressing #.
I want to execute an app when the control
exits from voicemailmain.
Any inputs?
Regards
Deepak
Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium
Zaptel cards?
Matteo Brancaleoni wrote:
Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto:
Klaus-Peter Junghanns [EMAIL PROTECTED] said:
we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN.
One thing I'd
When dialing out, will a call be established significantly faster by an
ISDN adapter such as an Eicon Diva server compared to an analogical FXO
such as Digium's X100P ?
signature.asc
Description: This is a digitally signed message part
There seems to be some trouble with either the maillist or my client. I
haven't received any of the posted replys on this topic, but found the
replys through the asterisk.linkx.net search engine. But anyways here is the
reply on the mail from: James H dot Cloos Jr. cloos at jhcloos.com
If it is
Hello all,
I wanted to know if is there a way to see which of my 4 g729b license
is registered in one specific Asterisk box. Is that possible? I could
not find any registration record on my box to compare with the
license...
best regards
Osvaldo
___
Am Mo, 2004-02-16 um 14.39 schrieb Jean-Marc V. Liotier:
When dialing out, will a call be established significantly faster by an
ISDN adapter such as an Eicon Diva server compared to an analogical FXO
such as Digium's X100P ?
Yes, ISDN uses digital signalling so call setup times on the last
Hi
Has anyone go the 30VIP phone to work with asterisk?
If so how good us the usability of the Cisco 30VIP phone with asterisk either
using chan_sccp or Chan_skinny?
Thanks for your Help
Robb
--
Robert Boardman
Tel:01617737929
FWD:86263
___
I need to get PSTN connectivity for my asterisk server. Either IAX2 or SIP.
Does anyone have recommendations of carriers that provide US termination
and will work (doesn't have to be a supported platform) with Asterisk?
OH: And you need not recommend Galaxyvoice. I'm not waiting 72hrs for
Hi,
today's BRI thread showed, that the mailing list has a delay of about
an hour again. Is this still due to the Digium relocation, or is
something else going on with the list server?
cu
Reinhard
___
Asterisk-Users mailing list
[EMAIL
Hi
How can i configure Asterisk for proxing SIP/SIMPLE Messages when the target
is registered? How can the user retrieve the waiting-messages?
Thank you very much.
.G
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
When you start * from console use -vvvc and the number of detected licenses
will be shown when the g729 translator is loaded. Only why that I know of to
check this.
Wes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Osvaldo
Mundim
Sent: Monday, February
I was wondering if there was any easy way to direct incoming calls from
IaxTel to specific extensions without having to create a separate context
for each? It seems to be pretty strait forward with FWD, SIPPhone and other
SIP based services -- you just add /Extension to the end of the
On Sat, Feb 14, 2004 at 02:47:11AM -0500, Ryan Finnesey wrote:
Does anyone know where I can find some more info on the VoIP laws in the EU?
VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU
parliament), last time they looked at it a few years ago it wasn't
perceived to be
John,
I've recently used nufone and voicepulse, both with great results.
Thanks,
Chris Clifton
- Original Message -
From: John Fraizer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 16, 2004 9:13 AM
Subject: [Asterisk-Users] VOIP Carrier recommendations?
I need to
The problem is that I have 2 licenses of 8 channels. One is being used
in one of my boxes and the other one is not. What I want is to be sure
that the one which I will use in a new Asterisk box is not the one
which is being used...
Any suggestion?
regards
Osvaldo
On Feb 16, 2004, at 11:57
Hi,
First let me apologize if I sent this to the list twice.
Is it possible to kick a caller out of a queue after 5 minutes and goto the
next priority in the context where they were assigned to the queue ?
My desired result is that even though one agent is dynamically logged into the
queue and
Billy Huddleston [EMAIL PROTECTED] wrote:
I've been using Asterisk with a Cisco GW and ATA's.. I have it setup with
re-invites. When a call is first answered, the 1st second or so of the
conversation is stuttered, garbled, whatever you want to call it.. I believe
this is due to Asterisk shifting
Does anyone know where I can find some more info on the VoIP laws in the
EU?
VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU
parliament), last time they looked at it a few years ago it wasn't
perceived to be entranched enough to worry about, I suspect this will
change
[EMAIL PROTECTED] wrote:
Hello All,
Does anyone here have any experience with pingtel Xpressa hard phones? I am considering buying a couple. Already have Snom200s, but want something with better CTI and full duplex speakerphone.
Michael
Michael,
I used some Pingtel phones with * and
Steve Kennedy wrote:
On Sat, Feb 14, 2004 at 02:47:11AM -0500, Ryan Finnesey wrote:
Does anyone know where I can find some more info on the VoIP laws in the EU?
VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU
parliament), last time they looked at it a few years ago
The correct way to hide your callerid on a PRI interface is to set the
presentation indicator.
Some CO switches do a basic sanity check on the callerid they receive. If
you set the number string to empty
but the presentation indicator to allow the number they will replace the
number string by your
I know that during the Registration that the file /var/lib/va-certificate is
created. Maybe this will help, file is encrypted so it don't offer much
information.
Wes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Osvaldo
Mundim
Sent: Monday, February 16,
Tim Sailer wrote:
I picked up a GS 100 phone based on the overall good response I've heard
of these phones. One thing I'm fighting with, which I can't find any
info on, is a *real* bad local echo on the GS. The remote end doesn't
hear it, and all the docs I see about echocancel deal with hardwired
Linus Surguy wrote:
Does anyone know where I can find some more info on the VoIP laws in the
EU?
VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU
parliament), last time they looked at it a few years ago it wasn't
perceived to be entranched enough to worry about, I
hi everybody,
here is what I've done to make my asterisk* act as a LCR.
first, you'll have to install isdnrate (part of isdnutils) and get a recent
rate-??.dat (check rates4linux.sourceforge.net for that.)
to test isdnrate just try the following command:
lcr -o -b3 -l60
Mayby you missed my reply as well. Here it is again ...
When I need to hide callerid ( sip phones ), I will configure this in
sip.conf.
You need to include restrictcid=yes
for each user that needs to be hidden.
-- Pertti
Mickey Binder wrote:
There seems to be some trouble with either the
Title: Asterisk - Carrier Access Bank Ring through
I'm currently having a very odd problem with Asterisk. I have a 24-Port FXO Access Bank I, and everything appears ok on the T1. No alarms, no issues at all.
The channel bank seems to be answering the line immediately, and I'm getting some
Andy,
Thanks for the feedback! I ended up buying one phone on Ebay for
$202.50, which seems like a good price. It would be used in a very
small office under light load. Perhaps I can get it nto work suffient
for my needs. Otherwise, back to Ebay to sell the beast ;-)
Michael
On Mon, 16 Feb 2004
On Mon, 2004-02-16 at 06:15, Tomica Crnek wrote:
Does it work? I can't find if there is a config option in pppd config
to make pppd authenticate user against radius server.
It should, but that is really a question for a ppp list or FAQ.
First link from google for me was this...
On Mon, 2004-02-16 at 07:39, Jean-Marc V. Liotier wrote:
When dialing out, will a call be established significantly faster by an
ISDN adapter such as an Eicon Diva server compared to an analogical FXO
such as Digium's X100P ?
Analog, nothing logical there.
ISDN will be faster dialing out as
On Mon, 2004-02-16 at 08:10, Robert Boardman wrote:
Hi
Has anyone go the 30VIP phone to work with asterisk?
If so how good us the usability of the Cisco 30VIP phone with asterisk either
using chan_sccp or Chan_skinny?
Thanks for your Help
I have had mine working with the chan_skinny.
The problem with the Ofcom consultation as I see it is that it seems to be
regressive wrt to the position now being taken by the FCC. There are
probably not many more than 250,000 VoB users worldwide so now is not the
time to impose significant market constraints.
The new EU regulatory
Just to clarify this from a different direction, Oftel/Ofcom approach
these
things by say that they are 'technology neutral', i.e. as standard they
don't care how the service is delivered, it is the service that is
regulated
and not the delivery mechanism. This means in theory the rules for
hi everybody,
Acer ISDN-Surf, Billion Bipac ISDN, Trust PCI ISDN Modem, D-LINK DMI-128+
to name a few ;-)
anyone knows where to get one of theses cards (or any other based on the
HFC-S chipset) in germany?
my computer-trader maybe can get d-link's card but he don't know how long it
could take.
The FritzCard has CAPI drivers and does NOT provide zaptel timing.
The quadBRI PCI has zaptel drivers and does provide zaptel timing.
Am Mo, 2004-02-16 um 14.41 schrieb Master Abi:
Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium
Zaptel cards?
Matteo Brancaleoni
VoicePulse--- If I get 2
simultaneous calls?
For one local number from voicepulse, is it possible to get
simultaneous incoming calls?
Jeff Chen
www.mutualphone.com
Yahoo messenger ID: jeffcheny2k
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
For Sale:
Adtran 850 Channel Bank with Router, running latest firmware A.04.04.26
Four FXS cards (that's 16 FXS ports for 16 separate asterisk extensions).
This is a great channel bank for Asterisk, when paired with a T100P card.
Auction ends in about six hours, currently at $370.
Klaus-Peter Junghanns [EMAIL PROTECTED] said:
Yes, like any zaptel device it supports echo cancelation (in software).
Good.
You can get 2 quadBRI PCI for the price of 1 Eicon 4BRI-8M.
Only? ;-)
--
Cees de Groot http://www.tric.nl [EMAIL PROTECTED]
tric, the new way
thanks :)
From: [EMAIL PROTECTED] on behalf of Steven Critchfield
Sent: pon 16.2.2004 20:22
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ZapRAS + RADIUS authentication
On Mon, 2004-02-16 at 06:15, Tomica Crnek wrote:
Does it work? I can't find if there
Hi -
I have looked but can't find details of the impedance that the analogue
PSTN interface card X100P presents. Am I right in assuming it is a
resistive 600 ohm match? If so, is there anything I can do (in software
or hardware?!) to 'tweak' its impedence to the complex impedence
normally
What is the advantage of having zaptel timing?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter
Junghanns
Sent: Monday, February 16, 2004 11:34 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Need to interface to BRIs
The
Hello list,
I am attempting to upgrade asterisk on a production box. I have opted to set
INSTALL_PREFIX to /usr/local/asterisk-0.7.2 which is ugly (since it
makes /usr, /var, /etc directories in there), but I didn't want the new install
to overwrite my existing installation.
The new asterisk
Hi there,
Got my DID from VoicePulse. Very fast and quite cheap :) I configured the
iax.conf with the info they provided, I am getting a good connect to their
server, but when I try to dial my number I am seeing the following on the
console. Anyone got an idea? Oh I replaced the number below
I really should have toned down my response before. For whatever reason,
google gives different results to different people. One may only
experience this when they use someone else's computer to do well known
searches. I apparently have good google juice when it comes to linux
related searches.
I have a couple of cummsy user who always lose a call when the transfer
is not done properly ie due to dialing a wrong digit, etc.
My question is that is it possible to savage a failed call transfer?
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
smime.p7s
Description: S/MIME Cryptographic
On Sun, 15 Feb 2004 21:00:53 -0500, Tom Knox wrote:
Thanks Michael, VoicePulse does have local number, so I just provisioned
one :)
Now I am setting it up, no problem so far, the next question is. If I get 2
simultaneous calls on my inbound will one ring busy or will asterisk handle
this for
On Mon, Feb 16, 2004 at 07:38:15PM +, Iain Stevenson wrote:
The problem with the Ofcom consultation as I see it is that it seems to be
regressive wrt to the position now being taken by the FCC. There are
probably not many more than 250,000 VoB users worldwide so now is not the
time to
I have a cordless phone that causes this same thing to happen every time I plug it
into a digium fxs port. I have an old style tdm card and a new one, same results with
both. Don't know what it is about the phone that makes this happen; It works fine
plugged into a pots line. The phone is a
I know having it unplugged from the line will cause this, but it's not.
It's an X101P single port FXO card. Most of the time it works fine but
occasionally wigs out. In this case zttool shows a red alarm. Other
times I call into it and it answers but I just hear a buzzing sound. In
a day
On Monday 16 February 2004 15:52, Tim Petlock wrote:
What is the advantage of having zaptel timing?
There are a host of features, such as conferencing and music-on-hold
which require a hardware device for timing. Not just any device will
work (it has to generate exactly 1000 cycles per second).
On Mon, 2004-02-16 at 16:16, Tom Knox wrote:
Hi there,
Got my DID from VoicePulse. Very fast and quite cheap :) I configured the
iax.conf with the info they provided, I am getting a good connect to their
server, but when I try to dial my number I am seeing the following on the
console.
On Tue, 2004-02-17 at 02:42, dkwok wrote:
I have a couple of cummsy user who always lose a call when the
transfer
is not done properly ie due to dialing a wrong digit, etc.
My question is that is it possible to savage a failed call transfer?
What kind of transfer, what kind of telephony
On Mon, 2004-02-16 at 15:52, Tim Petlock wrote:
What is the advantage of having zaptel timing?
Music on hold, and meetme require some form of timing. Also it makes
your sip phones better able to deal with VAD(silence suppression) by
providing a throttle to the audio.
-Original Message-
Steven Critchfield wrote:
On Mon, 2004-02-16 at 07:39, Jean-Marc V. Liotier wrote:
When dialing out, will a call be established significantly faster by an
ISDN adapter such as an Eicon Diva server compared to an analogical FXO
such as Digium's X100P ?
Analog, nothing logical there.
ISDN
Hi All...
I am using Asterisk successfully in my small office as a fairly ordinary
PBX. I am quite happy with it.
I have a friend who needs to build a call center. The call center will be
used to take orders. I have two big questions.
First, can Asterisk be configured accept calls on a
Iain Stevenson wrote:
The problem with the Ofcom consultation as I see it is that it seems
to be regressive wrt to the position now being taken by the FCC.
There are probably not many more than 250,000 VoB users worldwide so
now is not the time to impose significant market constraints.
Why do
yes it does and g723.1 also, im about to buy it, i will be sending
feedback to the list as soon as i get my unit. I really like alot how it
looks, hopefully i will also love how it works:)
Miguel
On Mon, 2004-02-16 at 12:57, HQ wrote:
Miguel,
IPC5000 doesn't support G729 (8 kbps) (it only
if my question is stupid just ignore please.
using SIP is the communication between the IP phones and the asterisk
server secure encrypted ?
Kemal
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I have 3 Pingtel phones and have tested them since they were
prototypes. I have had no lockups or
weird problems with them on Asterisk. I will says this about them though:
These phones are BIG on features and extensibility through Java at the
cost of quality. It doesn't
take a lot of work to
Howdy.. I am building * on a barebones server,
running just the minimum config (no X, etc..)
when I build, I get this error, and I'm trying to
track it down. Has anyone ran into this before or
have a general idea?
gcc -shared -Xlinker -x -o pbx_gtkconsole.so
pbx_gtkconsole.o `gtk-config --libs
Do any of you know of a cost effect device that could be connected to an
Asterisk station port to provide room monitoring? I'm looking to
replace the wireless baby monitor we currently have, since there is too
much interference between our daughter's room and our room for it to
work effectively.
When I make a simple phone call from one Budgetone 101 to another, the
speech sounds slurred and slow, sort of like the person is talking under
water. Both phones and the Asterisk server are on the same subnet.
Both phones are configured to use the PCMU (ulaw) codec as first choice, and
the Voice
Dear all,
I'm new to the list and new to Asterisk, so please bear with me ;) I've been
googling the web but couldn't find my answers... my apologies if these have
been already discussed before.
Nowdays I'm interested in setting up some VoIP-based solution on our offices
and I think Asterisk is
You'll probably want to re-quant them to 8kHz, but there are quite a
few classical tracks available at:
http://hebb.mit.edu/FreeMusic/
-JimC
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
On Mon, Feb 16, 2004 at 04:51:08PM +, WipeOut wrote:
I am going to an Oftel meeting to discuss VoB regulation next week..
Hopefully this will help to see where it is heading..
Of course Oftel doesn't exist anymore, it's all Ofcom now ...
Steve
--
NetTek Ltd Phone/Fax +44-(0)20 7483
Well, since they restricted attendance to service providers and
representatives of consumer organisations I wouldn't be too optimistic for
a balanced outcome ;-)
Iain
--On Monday, February 16, 2004 4:51 pm + WipeOut
[EMAIL PROTECTED] wrote:
Steve Kennedy wrote:
On Sat, Feb 14, 2004 at
This is excactly what restrictcid=yes
does in sip.conf.
Eg. when it is used you'll see this in pri debug:
Calling Number (len=12) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (
E.164/E.163) (1)
Presentation: Presentation prohibited, user number passed network
Where can we get the Eicon Dive server card with BRI ?
I mean, what is the lowest cost supplier ?
I have heard that we need to get the version after 2.x since the
echo cancellation is only supported after 2.x.
Regards
DL
___
Asterisk-Users mailing
VP seems to be working fine for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of matt
Sent: Monday, February 16, 2004 9:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Got my DID, getting an error.
I'm getting the same but they say they are
I am also going to share this with the list since it may be helpful for other
Wisip/* users out there.
There are two tricks that I discovered in getting the Wisip to work with *. The
first thing is to go into the Wisip's web config and select the SW/Update
section. There will be an entry for a
Jacky wrote:
Hi All,
I which to use Daemontools to watch asterisk process.
EVIL! Asterisk fork's a new process shortly after starting (unless you
run with a console)
Find safe_asterisk in the contrib directory.
Jeremy McNamara
___
I have the following config:
Asterisk compiled with oh323 on a public IP
Grandstream behind a NAT
Aseterisk sending calls to a Nextone MSW H.323
My Grandstream phone registers to my * server via SIP fine. When I place a
call that goes from my * server to my Nextone
I have been experiencing hung up when answering incoming calls through
x100p.
NOTICE[1242768320]: chan_zap.c:4584 ss_thread: Got event 2 (ring/Answered)..
-- Executing Wait(Zap/1-1,1) in new stack
-- Executing Answer(Zap/1-1,) in new stack
-- Executing DigitTimeout(Zap/1-1.5) in new stack
-- Set
Hi,
-Original Message-
Acer ISDN-Surf, Billion Bipac ISDN, Trust PCI ISDN Modem, D-LINK
DMI-128+ to name a few ;-)
anyone knows where to get one of theses cards (or any other
based on the HFC-S chipset) in germany?
my computer-trader maybe can get d-link's card but he don't
Hi all.
If I want to use the * only as a GW to PSTN and allow only one external
proxy to place calls. how is the smartest way to do this ?
I dont want the world to be able to do invites only a specific IP,
in this case my proxy that handles all the users.
/Mike
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