Bedankt voor uw bericht.
Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u
geinformeerd over de omstandigheden en uw opties.
Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of
beantwoord.
Indien uw abonnement is overgenomen door KovoKs,
Bedankt voor uw bericht.
Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u
geinformeerd over de omstandigheden en uw opties.
Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of
beantwoord.
Indien uw abonnement is overgenomen door KovoKs,
Bedankt voor uw bericht.
Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u
geinformeerd over de omstandigheden en uw opties.
Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of
beantwoord.
Indien uw abonnement is overgenomen door KovoKs,
Bedankt voor uw bericht.
Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u
geinformeerd over de omstandigheden en uw opties.
Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of
beantwoord.
Indien uw abonnement is overgenomen door KovoKs,
Distribution channels aren´t being made to the public yet, other then direct from digium. Looks like they will be waiting 2 weeks before we hear anything else. (After VON) The questions you are asking, I dont believe have yet been confirmed by asterisk or digium. Though I am sure it is on there
Hi
I
can use now DISA settings like this one when I set E1 card connected directly
to Asterisk. In this way every call dialed with pass 29 will be accepted. I
have a billing which filters caller ID number and address calls to each account
with same caller ID number previously set
Hi
I can use now DISA settings
like this one when I set E1 card connected directly to Asterisk. In this way
every call dialed with pass 29 will be accepted. I have a billing which filters
caller ID number and address calls to each account with same caller ID number previously
set
Hi
Tks for your info.
I can t set that
exten = s,1,Gotoif($[${CALLERIDNUM} = 1130851536 ]?10)
exten = s,2,Goto(from-pstn,s,1)
exten = s,10,disa(no-password,from-internal)
to work ok yet. I don t know
what are those contexts to (from-pstn) and(from-internal).
I cant
Hi,
I am trying to create a
situation where I call the DID number which is 1140636249 and I receive a dial
tone to dial. I d like also to autenticate the number 1130851536.
I can see that asterisk
decode this number but I dont know how to authenticate this number only. This
is what I
HI
I am trying to establish a connection between ASTERISK and ALEPO but I can
not,
since you have reached to make them communicate can you help me with the
changes made to asterisk, in this way I will be able to check if the
problem is the same with my ALEPO .
I would appreciate every help
My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x
But I can't use a 2.4.26 for some security reasons...
Alain
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Armin
Schindler
Envoyé : mercredi 5 avril 2006 18:05
À : Asterisk
OOps
The correct answer is
My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x
But I can't use a 2.6.x for some security reasons...
Alain
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL PROTECTED]
Envoyé : mercredi 5
Hello All,
I am using incoming DIDs for the first time. I ll very happy if someone
can help me on that serttings ... I need to know how to answer calls
from IP 200.123.123.1 with credentials abc123456:123456 and I d like to
address to extention 29650 incoming calls from that number which is
of G.729 since we already have a VAD
frame at the end
Is that possible to disable
Asterisk info messages ? If so .. what file can I edit in order to turn this
off ?
Regards
Newton
___
--Bandwidth and Colocation provided by Easynews.com
Hi,
I
need to send RTP from asterisk to one IP and signalling to another IP. In this
case, can you help me to arrange that configuration on sip.conf
[]
type=friend
username=
secret=
host=
dtmfmode=rfc2833
disallow=all
allow=g729
Atenciosamente
Diretoria
OOPPS! Looks like someone just broke voipjet's tos
gw at adcomcorp.com gw at adcomcorp.com wrote on
Sat Nov 5 11:36:46 CST 2005
I tend to agree with you, my experience with Teliax has been decent,
and
i had a similar problem a while ago. I solved it by defining
externip=xxx.xxx.xxx.xxx in sip.conf. It tells the remote SIP
client where you are.
-chuks.
Original Message Subject:
[Asterisk-Users] Can't hear the callerFrom: Lane
[EMAIL PROTECTED]Date: Mon, March 21, 2005 11:53
Hello,
I am using voipuser.org service, and am trying to make a SIP call.
Everything seems to work fine, except I can't hear anything on my end.
When I make a SIP call, the other party can hear me, but I can't hear
anything. I am using asterisk + Digium TDM board with an FXO port
where
Hi!
I've been using asterisk for 1 year now, however since yesterday
something odd happened. From my office i dial extension 125 and it
wont work, it sounds as a busy tone, and in the x-lite gives
me call failed 404 not found, however if i dial from my house to
that same extension then it
ok, thanks for pointing that out...
Original Message Subject: RE:
[Asterisk-Users] bridging iaxtel calls to PSTNFrom: "Rich Adamson"
[EMAIL PROTECTED]Date: Mon, February 21, 2005 4:00
pmTo: "Asterisk Users Mailing List - Non-Commercial
ok, thaks for pointing that out...how can I turn off the HTML tags? I am
using a web based email client.
BTW, sorry if this has been annoying, it's not been on purpose.
Original Message
Subject: Re: [Asterisk-Users] bridging iaxtel calls to PSTN
From: Jens Vagelpohl [EMAIL
Hello,
two questions:
1: How can I open/enable network connection to
B?
scenerio:
I have 2 Asterisk servers, A and B, running Fedora Core1
on my local network.B refuses any network connection attempts from
A, i.e. I can't even telnet or FTPto B from A, but I canto A
from B. This makes B refuse
Hello,
two questions:
1: How can I open/enable network connection to
B?
scenerio:
I have 2 Asterisk servers, A and B, running Fedora
Core1 on my local network.B refuses any network connection
attempts from A, i.e. I can't even telnet or FTPto B from A, but
I canto A from B. This makes B refuse
Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing
context
;add function here to
Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing context
;add function here to
could you help me out with this? I have a posting on this list, bu
nobody has replied yet. Titled "why can't I make IAX calls between 2
asrterisk servers"? I'd appreciate.
-chuks.
Original Message Subject: Re:
[Asterisk-Users] bridging iaxtel calls to PSTNFrom: "Michael
Graves"
Hello,
actually I did, but nobody responded to that. So, here it is
one more time:
___
Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go
Hello,
Can anyone help with this please?
thx,
chuks
Original Message Subject:
[Asterisk-Users] Why can't I make inter IAX calls between2 Asterisk
serversFrom: [EMAIL PROTECTED]Date: Mon, February 21, 2005
11:04 amTo: asterisk-users@lists.digium.com
Hello,
two questions:
1: How
Hello,
I just started using asterisk, and have a question. I have setup two
asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1
FSX modules) and is connected to the PSTN. B has same, but is NOT
connected to PSTN. I want to configure B to call A via iaxtel, and
connect to the
Hello,
I bought a TDM400P, and intended to use it with my analog
phone, which is RJ11 ofcourse. So, the question now, how do I plug in
my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also,
since it's an 11B card, I also intend to bring in an analog line into
the RJ45, so i am still
]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]:5060;line=v8ppcao5
P-Key-Flags: keys=3
User-Agent: snom190-3.56i
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported
protection.
Info: http://copilotconsulting.com/sig
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Hello,
I am using Broadvoice for my outgoing calls with my Asterisk box.
Broadvoice is requiring my to apply a patch to my Asterisk. Instructions at
the following link.
http://www.broadvoice.com/support_install_asterisk.html
Step 1 is what I need help with, not sure on how to apply patch.
I
Title: Message
Thanks to Greg Hill
for pointing me to the 'sip debug on' cmd that helped me resolve the sip
connection problem!
The other issue I'm
trying to resolve is configuring outgoing calls. I need to configure outgoing
calls to use the FXO card in the PBX (zaptel device) via sip
Title: Message
Nevermind. Figured this out. I needed the following in
extensions.conf to enable outbound dial.
exten
= _9.,1,Dial(Zap/2/${EXTEN:1},70,Tt)
Thanks
-Original Message-From: Info
[mailto:[EMAIL PROTECTED] Sent: Tuesday, August 17, 2004 9:27
AMTo: '[EMAIL
Title: Message
Hello:
Hoping someone might
know how to resolve this (probably an easy one). I have one Asterisk PBX with a
single NIC and an FXO card with PSTN line attached, and one IP phone (Budge Tone
100) on the LAN.Via the phone Iget no dial tone, and dialing 9,
number doesn't allow
I have made no recent changes to the IAX2 config on my system. Today I
tried a 1800 call and got the below error. Not sure when this started
since only use 800 once in a while. Does anyone know if IAXTEL is
experiencing problems connecting to the 8xx gateway?
7 16:14:54 WARNING[147466]:
Tim,
It looks interesting.. Are you willing to release the source code?
Robert
Tim Sailer said:
On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote:
Since there's not too much out there, I decided to take about 2 hrs and
pound something into shape for a simple status for my * server.
Angel Gabriel said:
I have 5 BT phone lines coming into my office. We use four for
international calls, and one for local/mobile calls. We have just obtained
another call carrier, and now we would like to be able to make calls from
any phone to any carrier, without having to remember what
Paul Mahler said:
With record:
; Record voice file to /tmp directory
exten = 9000,1,Record(/tmp/asterisk-recording:gsm)
exten = 9000,2,Hangup
Is there a way to stop recording other than hanging up?
Thanks!
Press the # key.
Below is from my extensions.conf. It plays the
Comments are inline.
Robert
Jeroen Rikhof said:
Hello,
Can somebody give me some information about:
1. How stable Asterisk is?
My experience and from what I have read on the list is that it is very
stable if run on stable hardware and you don't mess with the program code.
If you mess with
John,
You are now advertising your EMEA company in your signature block. Maybe
I missed an email that explains the EMEA pricing and availability. Could
you please give an update via the list as to the status of your product
availablity, pricing and delivery times in Europe? The ordering
Soren Rathje said:
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 22, 2004 8:52 PM
Subject: Re: [Asterisk-Users] SIP extension busy when not available ??
Although the current logic does not require a sip phone to
, yes please...
Well, I'm about three weeks into my very first * installation (that sort
of
works), so basically any info/tips/tricks/word of advice is accepted
with
appreciation...
-- Soren
I use a macro to define the extensions. In this way I only have to enter 1
line per actual extension
Jim Sneeringer said:
Whenever an outside number is dialed, Asterisk says We're sorry. Your
call
did can not be completed as dialed. Please check the number and dial again
or call your attendant to help you. I have tried many configurations,
but
let me give the simplest: It fails when a
[EMAIL PROTECTED] said:
I like using whisper tones...
recored the file companyname_whisper.gsm and put it in
/var/lib/asterisk/sounds
Then add the lines to extensions.conf
exten = 0031,1,Dial(SIP/Recp|20|A(companyname_whisper.gsm)r)
In my implementation of this the file extension had
Christian,
Where is a good place to purchase your phones in Germany? I found a
distributor in the UK but maybe just am not looking in the right place for
Germany.
Thanks,
Robert
American Expatriate in Friedrichshafen (Grund oder Entschuldigung für die
englisch)
Christian Stredicke said:
Sorry,
Tim Sailer said:
I've looked, poked, and hoped, but I can't seem to make * understand
the difference between a SIP channel being busy or not being there.
Both come up as 'busy'. I would expect the unregistered SIP to be seen
as unavailable. Am I just missing something obvious, again?
Tim
^
Andy,
I would be interested in your Cepstral engine code.
Regards,
Robert
Friedrichshafen, Germany
Andy Powell said:
lo,
Is there a single central location for code and applications other than
CVS? I'm talking about code that can't/wont be included in CVS for various
reasons? Does the wiki
Feedback for the list. I compiled Andy's code. Installation went well
(except for me misspellng something in the dialplan) with no problems.
The Application works great. Will run down Brian's and give it a try too.
Robert
___
Asterisk-Users mailing
Martin said:
Hello.
I vaughely remember someone talking about an asterisk implementation at a
University in germany some months back.
Any other information ?
Regards...Martin
--
http://graphics.cs.uni-sb.de/VoIP/en/index.html
Some of those folks and also from the Uni Stuttgart hang out
Real Player is required. Excellent video/slide presentation.
http://graphics.cs.uni-sb.de/VCORE/recordings.html
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Rob Fugina said:
On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote:
In the mean time, I've seen references to bug #'s, here on the list and
in the CVS logs. I've yet to stumble across the bug tracking system,
though -- can you give me a nudge in the right direction?
Thanx,
Rob
I have compiled the zaptel library and zaprtc on a system that gives the
following from uname -a:
Linux fxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27 13:58:12 UTC
2002 i686 unknown
Makefile for zaptel had the following line uncommented:
#
KFLAGS+=-D__SMP__
When doing the make load for
Mike Nash said:
Hi
I'm trying to configure my Asterisk box to provide a simple sample
configuration. It's a mandrake 9.1 box, no cards except a sound card.
The
config I am trying to achieve is simply one server, with two SIP clients.
Two issues are cropping up - the first, when I start
of your termination into Brazil. We
have several Brazilian expatriates here in Germany that might be
interested in your service. Partially would be Asterisk using IAX2 and
others using SIP Phones. Can you please pass along additional info?
Regards,
Robert
Friedrichshafen, Germany
John Todd said:
Time to dump the Netgear router. That's an unacceptable answer for a
router vendor to say Oh, well, for this MAJOR protocol we're going
to simply corrupt those packets so they're unusable. What!?
JT
__
OR get an older one from
Kannaiyan Natesan said:
Have anyone tried to interface BT's Broadband Voice with asterisk?
Kannaiyan
___
No, and not sure of their rates but http://www.telappliant.com/ has good
rates, voice quality and is easy to interface to Asterisk.
Robert
Kannaiyan Natesan said:
Do they offers, free evening and weekend calls? I get from BT.
You can get a free 0870 number from http://www.speak2world.com but they
charge for it.
Kannaiyan
Don't think so but sometimes free isn't free. Depending on calling
patterns it might actually be lower cost
Info based on how I do it is imbedded below.
Robert
Larry Keyes said:
I've got two Grandstream phones talking to * and a X100P card going, so
that
I can make inbound and outbound calls via the PSTN, and calls from one
extension to another.
1. Is there an equivalent to the more command
be accessible from the dialplan when an EnumLookup is
returned.
Anyone want to take a swing at it? Otmar? :-)
JT
John,
Thanks for the info. I'll leave the code commented out in the dialplan.
If I put in the NAT SIP patch then will reenable it. Is an interesting
concept for some long snowy night
John Todd said:
United States:* +1-800-...
+1-888-...
+1-877-...
+1-866-...
via: Telesthetic/Local Exchange Carriers of Michigan
JOhn, Good idea on leaving the code in. I'll do that. Since IAXtel has
8xx dialing
Top posting(sorry) then imbedding the answers to your questions. Otherwise
doesn't make sense.
Thanks for your reply. Sorry it took a while to get the answers. I'm in
Germany and your email came last night just as I was headed to the rack.
Robert
John Todd said:
my sip.conf contains:
Looks like the list server is really lagging tonight. I found out some
more info so will just post it in a new email with the same subject.
I added: search = freenum.org to enum.conf and got a match (SIP
system) when doing the lookup Maybe I overlooked that in the
original instructions
John Todd said:
...
Ideas welcome for more text; I may have another timeslot with Allison
early next week in which there will be some leftover room for
additional words. Short phrases and meaningful sets of words for
existing applications are desired; please don't give me words for
apps
John Todd said:
The freenum.org project wants to use your trunks! The freenum.org project
is an ENUM parallel tree, which has as an eventual goal the distribution
of ENUM numbering in nations or areas which due to political or other
issues are not able to get secure, inexpensive, or
Chris Albertson said:
I'm looking for a service that will accept VOIP calls and
send them to the PSTN. Or, I should say _another_ service
that will do this. I don't need the other direction
Currently I'm using IconnectHere and it works, but I get
complaints of poor audio quality from the
Chandra said:
i also had the same problem temporarily i solved my problem with both
outside NAT. u can also do it if both inside NAT. * outside NAT and
Budgetone behind NAT simply doesn't seem to work. if u ever solve this
problem please let me know too.
thanks
cm
I am able to use my
Philipp von Klitzing said:
oHi!
Ladies and Gentlemen, can anyone please help and let me know what is
the way to start Asterisk automatically using a cronjob, thanks
http://www.voip-info.org/wiki-Asterisk+administration
Philipp
Guess maybe I don't leave my system running long enough for
admin said:
I work for an interconnect that sells 3com and NEC. When I made this
project my own and followed through to show my boss, he said, this is
going
to ruin our industry
If that is the case then so be it. Same with mp3s and the music industry.
Had they embraced the technology,
,
or call +1 505 830 1200 and please do leave good
information (name, phone number, what you ordered)
we don't always receive enough info to respond back
(missing phone numbers or complete names are common)
If you have any issue you can call my direct number at
+1 505
It looks like Mark and others have addressed the development/CVS issues.
We should let their plan be put into effect and give it a chance to work.
Regarding the email list: A number of people have suggested creating more
email lists. I think this is not a good idea because there will be even
Hi,
Do the callers in USA dialling from USA Telco lines always have to
prefix the CITY/AREA code with 1 in order
To successfully make a call to other USA destinations?
I have not been to USA (yet) :)
Ta
SJ
For comprehensive info by area code (and as pointed out it does differ
from
Philipp,
Good document, my comments are inline with the parts to which they apply.
(and yes, this was a top post, otherwise it wouldn't make sense.)
Robert
Hi there,
mostly based upon list postings I compiled a couple of administrative
suggestions on the Wiki page below. I'd be glad to have
Philipp,
Good document, my comments are inline with the parts to which they apply.
(and yes, this was a top post, otherwise it wouldn't make sense.)
Robert
Hi there,
mostly based upon list postings I compiled a couple of administrative
suggestions on the Wiki page below. I'd be glad to have
a USB LCD display in my case to display things such
as:
Answering
Caller ID Info
Current Context
Etc.
I am very new to asterisk (in fact, I won't even be getting my digium
hardware until the 15th), so I'm sorry if this question isn't up to par
with the other discussions going on. Does anyone
Check http://www.telappliant.com for their VoIP Starter kits or Telephony
Cards sections.
Robert
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello there,
.
for pointing me at a friendly/knowledgeable UK supplier of such cards.
Any advice would be greatly appreciated: once I have
Where can I find that Howto? I'm new to Asterisk and am looking for all
the
doc I can find.
TIA,
Eric
Eric,
You will find at at:
http://members.lycos.co.uk/wipe_out/asterisk/
Robert
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I'm trying to buy a new X100P but
http://shop.store.yahoo.com/bsdmall/wisifxoin.html
is failing to check the order
Anybody knows any other way to purchase it?
Isamar
Try http://store.yahoo.com/asteriskpbx/wildcardx100p.html
You won't get the whopping 95 cent discount from BSD Mall but
Hi there,
yesterday I came across the Vocera Communication Badge and now I'd like
to know if anyone here has played with that thing (or even just seen it
in real life), and if a price tag can be found for this device?
Too bad they don't use SIP... ;-(
http://www.vocera.com/
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Quality Survey :P
Reply-To: [EMAIL PROTECTED]
...
I have 2 of these phones and they work fine for my application. Granted
its not the most intensive use and definatly not the most critical users
Message: 11
From: Asterisk online forums [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P
Date: Wed, 24 Dec 2003 11:23:14 -0500
Reply-To: [EMAIL PROTECTED]
Brian,
...
We are looking now to improve GS products and start collecting
The phone powers up and I can make calls through my Asterisk gateway to
other endpoints. However the four leds under the keypad are permanently
illuminated and the backlight slowly flashes on and off. When I pick up
the handset there is a repeated tone before I get a dial tone.
I know it's
Still, there seems to be a you get what you pay for theme to many of
today's posts and this clearly applies to support on FWD. Naybe we should
remove the signature from * that enables FWD to identify * systems :-)
That certainly seems the case for today's theme... It is certainly the
right
Interesting! Surely it would be another greate project.
Happy christmas!
- Original Message -
From: Bob Knight [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 11:30 AM
Subject: [Asterisk-Users] time to build an open phone?
Open software seems to work.
Why
Yes,I often get the same result, but not always.
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 20, 2003 3:40 PM
Subject: Re: [Asterisk-Users] DIAX phone busy
Yes, I've tried that as well. When I dial 70 from another
Hello,everyone,
I encoutered some difficult with IAX when I run
the asterisk.
internet -- asterisk + NAT -- DIAX
my * box and NAT are at the same linux box which connecting to the internet
using ADSL. The box has two network cards and two IP address,such as
public
Question2:
If I dial the IAX2 user registed to my * inside my NAT,it will
success,but
if I dial other IAX2 user registed to my * in the internet (not inside
my NAT),I alway get the result:
== Everyone is busy at this time
Take care that there is an issue with DIAX and IAX2... after some
Hi!
I don't get why people always say dtmfmode=info mine works fine with
rfc2833.
bkw
Dunno. I tried rfc2833 first, and had exactly the same problem as
described below with voicemail (but only there). Info then worked just
fine (as obviously also confirmed by this user here
I tried again at runlevel 3 but to no avail.
I'm pretty sure I have sufficient horsepower since I'm running on a box
with
half gig memory and a speedy CPU.
burak
I run on a Pentium I /100 Mhz, 32MB RAM with RedHat 9.0 and have no
trouble with voicemail audio or Music On Hold. This is a
Today I deleted the files in the asterisk, libpri, zaptel directories that
are in /usr/src and did a new CVS checkout (not update). After doing
the make installs and starting asterisk the show version is the same
as before:
Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586
On Sat, 2003-12-13 at 16:41, Joe Dennick wrote:
I just updated yesterday, but I did a complete rm -Rf for all of the
following directories:
/usr/src/zaptel
/usr/src/zapata
/usr/src/libpri
/usr/src/asterisk
Then I did a new cvs checkout for all four of those items
On Fri, Dec 12, 2003 at 01:57:02AM -0500, Brian Capouch wrote:
John Brown (CV) wrote:
Hi List,
Just a quick note that we have cleared all back logs of Grandstream
product. If you have been awaiting shipment, its shipped. Everyone
should be getting tracking numbers shortly.
We
it's a firmware problem on GS, they are working on that but it seems its
not that simple to make volume higher on the speaker and echo go away,
anyway 4.26 seems stable for now and with many new features!
Miguel,
What are the new Features?
Robert
___
Is anyone other than me having trouble dialing out via IAXTEL? I havn't
changed my config files in weeks but seems that IAXTel calls (800 and FWD)
stopped working in the past week sometime.
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Yes, I've been having problems as well but had not taken the time to
diagnose
the problem. Just did some looking and it appears iaxtel.com has removed
the iax v1 support. iax2 seems to be working fine.
Rich,
That solved the outbound problem.. Thanks for the hint... 800 numbers are
accessable
Hello every one,
I have got a
H323 gatekeeper for testing. The informations are something like
this:
account code: test01
gk ip address:192.168.10.12
I don't know how to set it in the h323.conf or
oh323.conf, I have tried it for almost one day but I always got the error. Help
me please.
Hi,Lubo,
Thank you very much for your reply. I want to use the gatekeeper for
outbound call, but I really don't know how to use it in the extensions.conf
,I think there are something diffrence between the chan_h323 channel and the
chan_oh323 channel. A little example of extensions.conf would
Thank's Lubomir and Jeremy! It's working now. That's to say,I could dial
long distance call from MSN or NetMeeting now.
Regards.
frank
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003
Hey, surprise! Just discovered it on the web:
http://graphics.cs.uni-sb.de/~rainer/tour.jpg
Mark is going on tour!
Not sure if this is real info or just a JPG that someone created.
Is Stuttgart a definate date on the 30th? If so, where in Stuttgart??
Robert
Friedrichshafen
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