I am trying to send calls from an AS5300 to Asterisk via e1 and I get this
bit of information in place of routing information
Going to extension s|1 because of Complete received
Accepting call from '' to 's' on channel 1, span 1
Here are the relevant zaptel and zapata pieces.
span=1,1,0,ccs,hdb3
What's your extensions.conf and sip.conf for your Grandstreams look
like?
I'm not in my machine right now, but here's the relevant configs
Extensions.conf
[ext]
Ignorepat=9
Exten=>_9XX,1,Dial(zap/1,tTr,20)
Exten=>_9XX,2,hangup
[sip]
Include=>ext
Include=>parkedcalls
Sip.conf
Folks,
I've been following recent discussions regarding echo and echo training with
much interest, since it's a problem we've never been able to eliminate here.
We're facing two challenges presently, and they may be related (or not):
a) Cisco 7960s in the office here echo back to our staff, but
Here's a puzzlers for you expert Asteriskians:
Person A in Australia with Xten-Lite connects to:
Person B on an Asterisk server in Vancouver, Canada (B is on analog phone to
FXS port)
quality is fine.
Person B on an Asterisk server in Vancouver IAX trunks to
our Vancouver PSTN gateway (running As
Anybody figured out how to close a VoicePulse Connect! account? As bad
as their web site is at most other things, the notion of actually
closing an account doesn't appear to have even been contemplated.
-brian
___
Asterisk-Users mailing list
[EMAIL PR
> I know, I know, check the archives but I can't find an answer since the
> new cards are well, NEW!
>
> I understand the whole issue of expandability and flexibility of using a
> T1 card and an Adtran 750. FXO or FXS, you mix and match.
>
> With the new card offerings from Digium I can easily pu
- Original Message -
From: "Alex Lopez" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, May 03, 2004 5:37 PM
Subject: [Asterisk-Users] Number of Digium cards in one box...
How much overhead would the 4 port cards put on a system?? At what point
would the breakeven point be??
-
Your English is just fine. :)
What's your extensions.conf and sip.conf for your Grandstreams look
like?
What are your options in the GS config webpage for:
1) NAT transversal (and are you behind a NAT firewall)
2) Send Flash event
3) Send DTMF
Best regards,
Ryan Thrash
On May 3, 2004, at 8:
As best I can tell you remove your credit card info, cancel any phone
numbers you have, and turn off the automatic billing stuff and when your
account hits 0 your canceled.
Scott
- Original Message -
From: "Brian Cuthie" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, May 03, 20
I record ALL calls in and out of my house. If someone calls me too bad.
Its not illegal in my state.
bkw
- Original Message -
From: "C. Maj" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, May 03, 2004 4:48 PM
Subject: Re: [Asterisk-Users] Start recording during call by pressi
ooops, this should have been norvergence.com
--
Lance
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk
On Mon, 2004-05-03 at 13:18, Michael Sandee wrote:
> Steven Critchfield wrote:
>
> >On Mon, 2004-05-03 at 12:39, Michael Sandee wrote:
> >It also means he is probably running the "STABLE" version. Debian named
> >it stable only because they don't change it very often. If you want
> >something as n
On Mon, 2004-05-03 at 14:15, Brian Capouch wrote:
> brian wrote:
> > If you live in a one-party state you can record ANY call in or out ..
> > doesn't matter if the call comes from/to out of state or not. You are
> > within the rights of your state. I live in Oklahoma and I record EVERY call
> >
On Mon, May 03, 2004 at 02:15:09PM -0500, Brian Capouch said:
> Is there a list somewhere of "one party" versus (I assume) "two party"
> states?
Google for telephone recording law
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http://lists.digium.com
On Mon, May 03, 2004 at 07:39:06PM +0200, Michael Sandee wrote:
> The bus isn't wrong... debian is wrong. Like everything in debian... it
> ships with an old pci.ids
> (No flames intended... but still :P )
>
> replace yours with one from:
> http://pciids.sourceforge.net/
>
> And It *should* repo
On Mon, May 03, 2004 at 03:13:56PM -0400, Tim Sailer wrote:
> On Mon, May 03, 2004 at 07:39:06PM +0200, Michael Sandee wrote:
> > The bus isn't wrong... debian is wrong. Like everything in debian... it
> > ships with an old pci.ids
> > (No flames intended... but still :P )
> >
> > replace yours w
I had just about about sold a new asterisk phone system to a local
company when they called back asking if I could match a proposal from
Novergence.com. I haven't seen anything on paper but was told their
proposal was to provide a new phone system that would replace the
existing 8 line 12 extensi
[EMAIL PROTECTED] wrote:
Check 0001436 in the bugtracker. This was the original bug fix which
broke outbound calls. Additional work was done on this bug to fix a
problem with incoming calls (see marks comments at the end). Maybe you
got a CVS while this was being worked on? Maybe there is still
Steven Critchfield wrote:
On Mon, 2004-05-03 at 12:39, Michael Sandee wrote:
The bus isn't wrong... debian is wrong. Like everything in debian... it
ships with an old pci.ids
(No flames intended... but still :P )
If that was all, then it would have still showed up in the PCI bus just
li
(History: Getting my home asterisk system set up; one land-line,
multiple SIP trunks; goal is to have a wife-proof transparent phone
system)
Just wanted to let everyone know that I got sipgate.de working with my
asterisk system; relevant settings below:
--account--
Sipgate number 8001234 (change
brian wrote:
If you live in a one-party state you can record ANY call in or out ..
doesn't matter if the call comes from/to out of state or not. You are
within the rights of your state. I live in Oklahoma and I record EVERY call
in our out of my house with asterisk.
Is there a list somewhere of "
On Mon, May 03, 2004 at 07:39:06PM +0200, Michael Sandee wrote:
> The bus isn't wrong... debian is wrong. Like everything in debian... it
> ships with an old pci.ids
> (No flames intended... but still :P )
>
> replace yours with one from:
> http://pciids.sourceforge.net/
>
> And It *should* repo
Check 0001436 in the bugtracker. This was the original bug fix which broke outbound calls. Additional work was done on this bug to fix a problem with incoming calls (see marks comments at the end). Maybe you got a CVS while this was being worked on? Maybe there is still a problem?
Bill Doll J
On Mon, 3 May 2004, Joel Duffield waxed:
> I am trying to get a way to have * forward calls that are dialed to an
> extension, to end up at an extension on my old analog phone system.
> I will have 7 lines coming into * using the new Digium cards via PSTN,
> and then lines coming from * into the P
Well, I just figured out that 4 digit dialing plan is used on the other
end so if I just 4 digit extension, i.e. 95222, the call works out fine
for both the sip phone and the pots phone.
However, I still don't understand that when I dial full 7 digit number,
sip phones work but pots does not. I'l
On Mon, 2004-05-03 at 12:39, Michael Sandee wrote:
> The bus isn't wrong... debian is wrong. Like everything in debian... it
> ships with an old pci.ids
> (No flames intended... but still :P )
If that was all, then it would have still showed up in the PCI bus just
like you mention below.
It als
Hans-Henrik Andresen wrote:
> Hi,
>
> If I do this in extensions.conf
>
> exten => 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],40,rS(10))
>
> the line is cut of in 10 sec., thats fine, but in CDR I got dst as T,
and not 411.
>
> How can I handle this so I still get kicked of after 10 sec.
Lists wrote:
Have you found a solution yet? I am having the same issue. My account works
fine with the IConnectHere soft phone client but not with Asterisk. Inbound
lines work fine, outbound returns the same message.
CVS update took care of my problem wrt outbound calls; it must have been
the mis-
The bus isn't wrong... debian is wrong. Like everything in debian... it
ships with an old pci.ids
(No flames intended... but still :P )
replace yours with one from:
http://pciids.sourceforge.net/
And It *should* report it better... (Didn't verify)
Not that any of this matters... Just load the dri
Thanks for the update!
Michael
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, May 03, 2004 1:18 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
iconnecthere behind NAT, strange deal
Hi,
I had the same issue with i
On Tue, 2004-05-04 at 06:46, Nick Grindley wrote:
> Hi All,
>
> Many thanks to Marc who helped me with a previous Capi Dialout plan -
> however.
>
> What I now would like to be able to do is: -
>
> We have 8 msn's 383590, 383591 383592 etc.
>
> What I would like to do is set up an If Then E
On Mon, 2004-05-03 at 12:26, Tim Sailer wrote:
> Folks,
> I'm trying to install one of the new quad fxo cards remotely. I know
> the existing machine was too old to have a PCI 2.2 bus, so I had my
> helper at the other end try a few boxes that were sitting on a shelf
> with the new card and a Kno
sure.
use * with:
IAX2 for sending voice via WAN (I suppose is internet)
and then for ISDN you can:
if is PRI , get 2 digium cards
if is BRI , get zapbri cards
matteo
Il lun, 2004-05-03 alle 19:41, Patrick Stuckenberger ha scritto:
> Hi list,
>
> is it possible to create something like a ISDN-W
Hi list,is it possible to create something like a ISDN-WAN-WAN-ISDN
bridge?
We have to change our location, but our number and the telephone system
should shoulb stay the same.
kind regards,Patrick Stuckenberger
___
Asterisk-Users mailing list
[EMAIL P
On Mon, May 03, 2004 at 11:13:17AM -0500, brian said:
> If you live in a one-party state you can record ANY call in or out ..
> doesn't matter if the call comes from/to out of state or not. You are
> within the rights of your state. I live in Oklahoma and I record EVERY call
> in our out of my ho
Folks,
I'm trying to install one of the new quad fxo cards remotely. I know
the existing machine was too old to have a PCI 2.2 bus, so I had my
helper at the other end try a few boxes that were sitting on a shelf
with the new card and a Knoppix cd. He found one that reported the
card as the Tiger
Hi,
Try latest CVS. There was an auth sip bug fixed on Saturday.
http://bugs.digium.com/bug_view_page.php?bug_id=0001533
Hope this helps.
Bill Doll Jr
Mark Turner <[EMAIL PROTECTED]>
Mark Turner <[EMAIL PROTECTED]>
Sent by: [EMAIL PROTECTED]
05/02/2004 05:24 AM
Please respond to
[E
I have two units running 1.0.35a working just fine.
On Mon, 2004-05-03 at 10:08, Victor Perez wrote:
> I just tried to upgrade my sipura to firmware 1.0.35a and now I can't connect to it.
> It still works but any connection to ports 23 and 80 makes it reboot. Even the flash
> tool makes it to
On Mon, 2004-05-03 at 12:05, jimfl wrote:
> So does this mean you could get direct RTP steams between a SIP client and
> a IAX2 client? What about inband/out of band DTMF issues?
IAX/IAX2 does not use RTP.
--Eric
--
Useful Asterisk Docs (BOOKMARK THEM!):
http://www.digium.com/index.php?menu=do
Hi!
> What I would like to do is set up an If Then Else type statement along the
> following lines: -
>
> If extension 7957 Then
> Dialout on Capi msn 383590
Create a macro in extensions.conf:
exten => s,1,AbsoluteTimeout(${TIMEOUTABS})
exten => s,2,NoOp
exten => s,3,GotoIf($[$[${CALLERIDNUM} =
On Mon, 2004-05-03 at 12:05, jimfl wrote:
> So does this mean you could get direct RTP steams between a SIP client and
> a IAX2 client? What about inband/out of band DTMF issues?
IAX doesn't use rtp and therefore it couldn't do it either. All DTMF
should be OOB to be reliable.
--
Steven Critch
Hi,
I had the same issue with iConnectHere as well as FWD (authenticated). If you are running CVS -head (4/26 - 5/1) i would suggest getting the latest CVS. I worked with Mark over the weekend to resolve this bug.
http://bugs.digium.com/bug_view_page.php?bug_id=0001533
Hopefully, this will r
I just tried to upgrade my sipura to firmware 1.0.35a and now I can't connect to it.
It still works but any connection to ports 23 and 80 makes it reboot. Even the flash
tool makes it to crash when trying to connect. Anybody else experiencing this problem?
Regards,
Victor Perez
___
- Original Message -
From: "Jeremy McNamara"
To: <[EMAIL PROTECTED]>
Sent: Monday, May 03, 2004 12:48 PM
Subject: Re: [Asterisk-Users] Asterisk remains in the media path
> brian wrote:
>
> >Can't do it because you are changing from one technology to another.
> >
> >
> >
>
> Actually
Hey,
Someone told me an open source SGCP gateway was created for the Asterisk
project. I'm looking for a little more information.
I have two VG248s that I'd like to attach to my VoIP network; however,
Cisco's documentation seems to indicate that Cisco CallManager is
required for these things
brian wrote:
Can't do it because you are changing from one technology to another.
Actually its cuz chan_h323 sucks like that.
Jeremy McNamara
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul Berger
Sent: Monday, May 03, 2
Thanks a million.
On Mon, 2004-05-03 at 12:04, Steven Critchfield wrote:
> On Mon, 2004-05-03 at 10:32, Joseph wrote:
> > Thanks for the answer.
> >
> > I only joined the list 4 days and did not mean to ask a reduntant
> > question.
>
> This is why the mailing list is archived by digium and ind
If you live in a one-party state you can record ANY call in or out ..
doesn't matter if the call comes from/to out of state or not. You are
within the rights of your state. I live in Oklahoma and I record EVERY call
in our out of my house with asterisk.
bkw
___
On Mon, 2004-05-03 at 10:32, Joseph wrote:
> Thanks for the answer.
>
> I only joined the list 4 days and did not mean to ask a reduntant
> question.
This is why the mailing list is archived by digium and indexed by
google. Redundant questions should be easily covered. I'm not sure but
the wiki
Le lun 03/05/2004 à 17:34, brian a écrit :
> Can't do it because you are changing from one technology to another.
Thanks for your answer.
H323 and MGCP are supposed to stay on the call control level, why isn't
it possible to open RTP channels between the terminals then?
Again, thanks,
Paul
__
Jeremy,
In the realm of US law, there are no definates. Before going any farther,
I AM NOT LICENSED to practice law or provide legal advice in any way, form,
or manner. That being said, talk to someone who is licensed to do so. And
in this case, make sure that person is or has access to a
t
Hi all,
I'm using txfax and rxfax to send and receive faxes. I find that txfax performs very
well when sending to an old fax machine. However, when trying to send to a Mac (OS X)
that accepts faxes, I get a "Trainability test failed" message on the * console (full
console log below).
I'm OK
Can't do it because you are changing from one technology to another.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Paul Berger
> Sent: Monday, May 03, 2004 10:29 AM
> To: Liste Asterisk
> Subject: [Asterisk-Users] Asterisk rema
Thanks for the answer.
I only joined the list 4 days and did not mean to ask a reduntant
question.
I will see if I can get it fixed.
I assume turning callerid would not make it wait for that,
so the shortest amount of time might be 1 to 2 rings then?
Any tips on debugging callerid problems?
O
Hi all,
Just a quick question: I have an H323 terminal and some MGCP phones
connected to *, and when they call each other * remains in the media
path no matter what (while I'd like to have the RTP stream directly
between the phones).
- mgcp.conf has canreinvite=yes
- extension.conf doesn't contain
On Mon, 2004-05-03 at 10:04, Joseph wrote:
> I have 2 X100P cards that I am using to handle voicemail,
> but I have problem.
>
> It takes about 3 to 4 rings before they pick up.
>
> Anyone have any idea why it takes so long to pickup and answer?
>
> Or is there a way to control this?
Do a minor
Have you found a solution yet? I am having the same issue. My account works
fine with the IConnectHere soft phone client but not with Asterisk. Inbound
lines work fine, outbound returns the same message.
Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behal
I will double check.
How much cpu does the MeetMe feature need
per user?
Or does it depend on how they connect?
On Mon, 2004-05-03 at 11:54, James Sizemore wrote:
> Check the duplex on your ethernet conection on both the Cisco and the
> Asterisk box. Make sure neither are half duplex.
>
> Jose
I have 2 X100P cards that I am using to handle voicemail,
but I have problem.
It takes about 3 to 4 rings before they pick up.
Anyone have any idea why it takes so long to pickup and answer?
Or is there a way to control this?
--
respectfully, Joseph
__
I´m using * 0.9.0 and have a X100P connected to my analog PBX. I can´t
detect hangups on this line, so I turned on busydetect=yes in
zapata.conf. I also have busycount=6.
While the line is connected to the PBX, I can never detect busy and the
line hangs at the end of every call. If I connect t
I don;t know if stealth really that important..
How many times have u hung up on a company when they told you that your
call may be recorded for quality and training purposes??
I hear that from every company I call.
Most people brush off the message forget about it befor they even talk
to someon
Check the duplex on your ethernet conection on both the Cisco and the
Asterisk box. Make sure neither are half duplex.
Joseph wrote:
What codec should be used to connect a * box to
a cisco router which has a t1 with 24 trunks coming in?
My router voip dial plan looks like this:
dial-peer voice 2
Zac,
Thanks for the input. This would cover it, however it is not stealth.
In some cases, you may want it to be stealth. Again, my state allows me
to do this, but some states do not. I've done a little searching and
could not find an answer. Basically, to simplify the question: When is
it lega
Why don't you have a
"This call may be recorded for quality assurance"
when someone calls in
That provides notification.
Zac
On Mon, 2004-05-03 at 09:01, Jeremy Hall wrote:
> Does anyone know how these laws apply in interstate calls? For example,
> I am in a One-party consent state. This
no I usually have 2 to 3 calls going down a full data T1(only voice
data) and I get this message and 2 sec later calls are dropped. we look
at our bandwidth for that time and we were no where near full
utilization.
On Mon, 2004-05-03 at 13:58, brian wrote:
> 1. Its not an error.
> 2. It's a warni
Does anyone know how these laws apply in interstate calls? For example,
I am in a One-party consent state. This means I can legally record any
telephone I am a part of, without notifying any other party. Say
someone from Florida or another all-party state calls me, or I call
someone in Florida.
1. Its not an error.
2. It's a warning.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Justin Carlson
> Sent: Monday, May 03, 2004 3:21 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] IAX2
>
> we are getting these err
Title: Message
Looks like you might have a hardware
issue.
bkw
-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Jim O'Brien
Sent: Monday, May 03, 2004 8:36 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Error
building asterisk-0.9.0
Title: Message
I am trying to build
asterisk-0.9.0 on 533MHz 160MB Redhat Linux 9.0 machine.
I have followed the
instructions to build asterisk.
Building zaptel and
libpri seemed to go well (lots of messages but nothing that indicated an
error)
However, when I do
the make clean ; make
we are getting these errors too which cvs was it fixed in ? we just
upgraded to cvs-stable from friday to see if that would help.
On Sun, 2004-05-02 at 21:45, brian k. west wrote:
> I think this was fixed in CVS-HEAD because I do not see that message
> in the src at all while looking to see if t
Someone wrote:
> The "BT CD50 and soldering iron" plan is looking more and more like the
> one I'll be going with for now
If you don't fancy using a soldering iron to read UK CLI there's a mod to *
that my colleague, Robb Boardman, uses. By placing a certain model of Hayes
or Pace modem in pa
Joe Baptista wrote:
agreed - you see alot of business fluff - but the technicals are very
important and on many of these ventures they fail to include them.
As far as I'm aware they are providing an internet exchange peering
point for voip providers, and to get access to their enum zone you need
I am trying to get a way to have * forward calls that are dialed to an
extension, to end up at an extension on my old analog phone system.
I will have 7 lines coming into * using the new Digium cards via PSTN,
and then lines coming from * into the PSTN lines on the analog system.
So that if for exa
Hi klaus-peter,
I thought I fixed this error... but
when ever I pickup the phone before I dial the number (the sitution I got
the former descibed problem fixed with overlapdial=yes) I can dial an
extension but I cannot send any furhter digits so voicemail and early
b3-connects with chan_capi
I've been running * for eight months in production mode without the
init.d/zaptel script in place. Didn't know 'make config' from within
the zaptel src directory even existed, and have never seen/heard anyone
even mention that before. Its been running fine with a pair of x100p's,
however the system
On Mon, May 03, 2004 at 09:28:56PM +1000, Adam Goryachev wrote:
> Firstly, the problem...
>
> Ever since I installed and setup asterisk, I have had various problems,
> initially it was echo caused by the ISDN (isdn4linux) card I was using.
> So, I upgraded to the X101P from digium. I still had ech
Hi Adam,
what is your echocancel setting in zapata.conf for the PRI spans?
I once noticed this distorted sound by using echocancel=256 (using
mec2.h for echo cancelation).
How about echocancelwhenbridged and echotraining?
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Hi All,
Many thanks to Marc who helped me with a previous Capi Dialout plan -
however.
What I now would like to be able to do is: -
We have 8 msn's 383590, 383591 383592 etc.
What I would like to do is set up an If Then Else type statement along the
following lines: -
If extension 7957 The
Damn, I forgot to describe the actual problem. Basically as someone I
spoke to today described it, it sounds like you have one of those new
digital pbx systems... In more detail, when he spoke, he heard his voice
come back, but distorted. The louder the sound he made, the louder he
heard himself (d
Hello,
Can we see your dialplan related to that ?
On Mon, 2004-05-03 at 13:40, Osvaldo Mundim wrote:
> Hello,
>
> I'm using an AGI program written in C to manage incoming calls to some
> extensions. Its being used for a small call center (20 people).
>
> When the call comes in, the caller can
Hello,
I'm using an AGI program written in C to manage incoming calls to some
extensions. Its being used for a small call center (20 people).
When the call comes in, the caller can listen the directory menu and
then dial the extension. The AGI program is called and get one of the
available ext
Firstly, the problem...
Ever since I installed and setup asterisk, I have had various problems,
initially it was echo caused by the ISDN (isdn4linux) card I was using.
So, I upgraded to the X101P from digium. I still had echo, so I figured
it was also caused by the ATA186 (cisco) I was using. So,
Il 21:38, venerdì 30 aprile 2004, Philipp von Klitzing ha scritto:
> Hi!
>
> > I try to connect an MGCP device(Terayon) to asterisk. I have found many
> > example BUT the Terayon always return error 510 ! "Verb:'510'
> > Identifiers :'2' Endpoint: 'Error' Version'(null)'
>
> 1. Which version of Ast
Andres wrote:
I think the username/secret items in sip.conf are busted. A quick
ethereal trace shows that when placing an outbound call to another
provider via SIP, * is not using the username defined during the
authentication challenge, instead it uses the username of the phone
placing the ca
I had gotten one of these off of eBay. It had one FXS card in it.
A quote from Vina for an additional FXO card was $1100.
And the unit I had was vibration sensitive. Flick the case with my finger and
the T1 would go offline for a few seconds.
Well, I got it for $150. (Get what you pay for.)
right, so far, here is what I've done:
I succeed in take in a new variable the UUS1 field sent with the connection
request for incoming calls. It was quite simple afterall... (I just had to
find where the data CMSG->Useruserdata is coming in chan_capi.c)
Now I would like to know where this f
forget it... seems to work - no idea what was/is wrong.
- Original Message -
From: "FastJack" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, May 03, 2004 11:38 AM
Subject: Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a
> hi klaus-peter,
>
> yepp... with overlapdial=
hi klaus-peter,
yepp... with overlapdial=yes (almost) everything works great, again.
one problem is left... touchtones are not working anymore so I can't use
voicemail-system, parking and stuff.
thank you for your help.
...bye
thorsten
- Original Message -
From: "Klaus-Peter Junghanns"
> Whenever we have needed any kind of Carrier Access
> (Adit) equipment, we have used Suntel Data. We found
> them on eBay and have been buying from them ever
> since. They have always been good about fast shipments
> and provide good used pricing on equipment.
> http://www.sunteldata.com/
I've al
On Mon, 2004-05-03 at 02:30, Jon Brandon wrote:
> On Mon, 3 May 2004, Steven Critchfield wrote:
>
> > On Mon, 2004-05-03 at 01:06, Jon Brandon wrote:
> > > * We will start with 6 PSTN lines
> >
> > If you are going to start with 6 lines, you should decide how soon you
> > might upgrade. You the
Hi everybody,
I have a MTA from Terayon that I try to make run with Asterisk using
MGCP channel.
The device is running with MGCP 1.0 NCS 1.0
Each time Asterisk try to send a Request (Request Notify, Audit
Endpoint) the device returns error 510 "Protocol Error"
Does anybody have already me
On Mon, 3 May 2004, Steven Critchfield wrote:
> On Mon, 2004-05-03 at 01:06, Jon Brandon wrote:
> > Hello,
> >
> > I'm about to install asterisk as the PBX at a location that my company has
> > just moved into and I would like to get some comments and advice on the
> > installation. I am new to
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