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, kij
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, kij
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, kij
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, kij
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 min
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
[f
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-inter
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 you
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 avri
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 Use
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
11406
frame 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
Direto
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 get
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 11:5
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 wil
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 P
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
Discussion"iaxtel is
not working and hasn't
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: Ho
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
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,
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 conti
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 cont
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 FTP to B from A, but
I can to A from B. This makes B refu
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 FTP to B from A, but I can to A
from B. This makes B refus
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
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 PS
nch=z9hG4bK-436npujwg1i0;rport
From: "snom_01" ;tag=jdx5841oim
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact:
P-Key-Flags: keys="3"
User-Agent: snom190-3.56i
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER,
gned for your 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 ha
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:
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 con
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 I get no dial tone, and dialing 9,
doesn't allow me to
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]: chan_i
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 * serv
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 d
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
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 wit
I can send it to you from home
>> tonight.
>>
>> Robert
>>
>
> Thank you, 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
> w
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
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 procedu
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 whe
Matt McIntyre said:
> I am interested in subscribing to a service that will let me dial the
> PSTN in Ireland and am interested in what the community thinks about who
> has the best services available. I would prefer to purchase time in
> blocks of minutes or pay as I go in lieu of having a monthly
[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 ext
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?
>
> Ti
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 l
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 wi
Real Player is required. Excellent video/slide presentation.
http://graphics.cs.uni-sb.de/VCORE/recordings.html
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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
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,
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"
t;
Daniel,
I would be interested in the details 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?
R
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 s
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 on
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 low
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
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
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 dia
lready a "preference" factor built into NAPTR records
> that should 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
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
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:
>>
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 fun
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
> ap
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 fro
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 t
t; > > > > Mike received email replies on 3-Dec and 17-Dec advising him
>> > > > > on his order.
>> > > > >
>> > > > > Mike ack'd those emails.
>> > > > >
>> > > > > This is the first time we have h
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 technol
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
> Morning All,
>
> I have created some virgin forums that I think may relinquish the mailing
> lists from major burdens. Everything is .001 in version and I need help.
>
> I need some advice as far as images and content. I know the project is
> opensource but is content and graphics? If not can
> 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 comprehensi
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
more
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 h
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 h
nyone other than home users, but I
> would like to use 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
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: onc
John wrote:
> Hi
>
> This is hard work :) I have read the Asterisk Handbook, BudgeTone User
> Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource
> Pages
> and more.
>
> I am not a linux newbie but am new to Asterisk. I have failed to find any
> docs that explain how to get a very
> 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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http:
>
> 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
> Is that FCC sticker on the back of the phone for real?
>
> A customer could not use his computer while talking on his GS BT102 phone.
> The customer was using a major name wireless keyboard/mouse with his pc.
> The keyboard/mouse stops working if the GS phone is too close.
>
> --
> Bob Knight
> [
> Hallo.
>
> I am living oin Germany and having two ISDN BRI Lines available. Capi
> driver!
>
> I need a Sip Gateway and a H 323 Gateway.
> About H.323, there should be a full implementation of H.450.
>
> Which software is available that gives me a Sip and a H.323 Gateway to
> enter
> my PSTN with
> 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/
> http
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.
> > 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
r
>
>
> 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
>> 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 st
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
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 anoth
> >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.
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 IP:211
> 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 o
> 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 t
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 install"s 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
> 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.
> 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
>> 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 number
>
> 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
acces
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