With your qwest thing. I don't know of anyway possible. Possibly a 3 way
type call. But I believe the call would drop when they hang up.
That would be a cool feature/something to se working.
Michael
- Original Message -
From: "Ryan Parlee" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent
1) This should be really easy, but I can't seem to get it working; I want
the caller to be able to dial an extension while Asterisk is doing a Dial
command. Is this possible?
2) While I'm at it...has anyone called Qwest lately? When they transfer
you, they can stay on the line the entire time,
Folks,
I'm experimenting with bringing multiple (8) analog lines from our local
telco into a Carrier Access ADIT 600 channel bank with an FXO module, then
having this talk to Asterisk via the T1 TDM controller on the ADIT and a
TE405P card.
I don't know if this will work well (ie: give me decent
> Is there any word on the availability of the FXO cards for the TDM400P?
> I have an application that would benefit. If it has been dropped please
> let me know.
Word has it that they should hit distributors in the next week or perhaps
two. One caveat -- they do not have FCC certifications yet.
If you have a new enough version of the IAXy firmware on the IAXy, then it
will automagically be upgraded as soon as * sees it has an old firmware
(via the IAX protocol) --- if you don't have a new enough version, digium
has to do it by what I've heard
Sam
-Original Message-
From: [EMAIL
Not to sound like a broken record but..
Is there any word on the availability of the FXO cards for the TDM400P?
I have an application that would benefit. If it has been dropped please
let me know.
Thanks
Gene Kochanowsky
___
Asterisk-Users mailing
> The robbed bit T1 has 2 signalling bits, but the usually do the same
> thing, so its really like having just one signalling bit. However, the
> timing of changes to that signalling bit can follow one of several
> patterns. You will see the terms immediate start and delayed dial used,
> as well as
Hello,
Use openvpn http://openvpn.sourceforge.net this will solve your woes.
Thanks
-Matt
- Original Message -
From: "centauri Star" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, April 15, 2004 7:53 PM
Subject: [Asterisk-Users] ATA 186 SIP behind XP Dynamic IP Firewall to
Hi all
I was tryed to connect to mysip.ch scs_client by siemens that isn't
works well.
Does anyones knows to work H/W or S/W applictations in asterisk SIP
videosupport?
Regards
mack_jpn
___
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[EMAIL PROTECTED]
http://lists.
Is it possible to set up the following?
public IP Asterisk Server to ata 186 behind a XP server firewall.
I think my biggest problem is that I don't know how to make XP forward the
RTP port to the private ata address.
I would put up some configs, but was hoping that someone one who has this
w
Mike Machado wrote:
cvs HEAD did infact fix the ringing problem. Thanks Eric!
As I said, CVS STABLE also has the fix as of this afternoon.
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUB
Mike Machado wrote:
cvs HEAD did infact fix the ringing problem. Thanks Eric!
I have another question for all you T1 buffs out there. The T1 I am
working with goes into our local phone switch (Excel switch). Currently
we are using E & M Wink signaling. The problem is we cannot set callerid
on the
Andrew Kohlsmith wrote:
When we had a MCI ct1, they couldn't send us proper supervised hangup on
a loopstart encoded DS0. They claimed it to be a problem with the
software on their switch. Their solution was to switch to groundstart.
Our end solution was to drop them and switch to Telcove(formerly
Thank you.Simon Brown <[EMAIL PROTECTED]> wrote:
You could try these:
voiptalk - www.voiptalk.org
sipgate - www.sipgate.de
Simon
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McMillinSent: Thursday, 15 April 2004 15:29To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Most Re
cvs HEAD did infact fix the ringing problem. Thanks Eric!
I have another question for all you T1 buffs out there. The T1 I am
working with goes into our local phone switch (Excel switch). Currently
we are using E & M Wink signaling. The problem is we cannot set callerid
on the outbound side. My mi
--On Thursday, April 15, 2004 6:43 pm -0400 Steven Kokinos
<[EMAIL PROTECTED]> wrote:
Actually, after rebooting my machine music on hold started working
properly. Not sure what the issue was. As for ztdummy, I am having a more
substantive issue with that, which is keeping me from getting meetme
Hello. My installation does not have a timing source and music on
hold works just fine.
My guess is that you forgot to put mp3 files in your /var/lib/asterisk/mohmp3
directory.
My /etc/asterisk/musiconhold.conf file looks like this:
;
; Music on hold class definitions
;
[classes]
default
Just wondering if anyone has used the Sipura SPA-2000 Voip to Single line
phone adapters?
These would be used for Fax machines and our Security system.
http://www.voxilla.com/shop/index.php?action=item&id=3&prevaction=category&previd=2&prevstart=0
Kyle
_
On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
> Explicitly answer the line. If that doesn't handle inband audio, there
> is a r flag to dial. This was discussed very recently.
This must be a different problem, because neither of those solutions
worked.
zapata.conf sends call to fixu
Actually, after rebooting my machine music on hold started working
properly. Not sure what the issue was. As for ztdummy, I am having a
more substantive issue with that, which is keeping me from getting
meetme working.
while ztdummy compiles cleanly, i can't actually get it to load
properly.
Fixed in CVS STABLE around 2pm CDT today. It's been fixed in CVS HEAD
for a while.
Mike Machado wrote:
When people call into my * box over the T1 interface, they get no ring
tone. It rings the SIP phone and when the SIP user picks up, both
parties can hear each other ok, its just the PSTN user c
On Thu, 2004-04-15 at 17:18, Mike Machado wrote:
> When people call into my * box over the T1 interface, they get no ring
> tone. It rings the SIP phone and when the SIP user picks up, both
> parties can hear each other ok, its just the PSTN user calling in hears
> no ring. What could be causing th
When people call into my * box over the T1 interface, they get no ring
tone. It rings the SIP phone and when the SIP user picks up, both
parties can hear each other ok, its just the PSTN user calling in hears
no ring. What could be causing this?
I tried setting immediate to yes in zapata.conf, bu
I have been having a major problem with after some installations of Asterisk
about every 3rd one the CLI will come up in some strange looking greek
letters. This problem does not happen all the time but once it happens I
was not able to clear it up.
Well with the help of a unix/linux expert we ha
Hi!
> Only Grandstream phones appear to be affected. All phones affected
> have been behind a coned NAT, running firmware 1.0.4.39 with STUN
> enabled. The hangup only occurs in dialogs with CSeq set to '0'.
Ok, I'll watch for that as well since I upgraded my desk's Grandstream to
1.0.4.54 an h
I wrote:
> In article <[EMAIL PROTECTED]>,
> Steven Kokinos <[EMAIL PROTECTED]> wrote:
> > i've been searching the archives but can't find anything substantive on
> > this. most of the music on hold documentation discusses integrating
> > with zap hardware, but i am trying to send it across a sip
In article <[EMAIL PROTECTED]>,
Steven Kokinos <[EMAIL PROTECTED]> wrote:
> i've been searching the archives but can't find anything substantive on
> this. most of the music on hold documentation discusses integrating
> with zap hardware, but i am trying to send it across a sip channel.
Music on
I have a dlink dvg-1120s voip-router. I can make calls out from the router,
but when calling the router I got
-- Executing Dial("SIP/2010-b437", "SIP/2021|30|r") in new stack
-- Called 2021
-- Got SIP response 404 "Not Found" back from 62.79.78.74
-- SIP/2021-473b is circuit-busy
Victor Perez wrote:
I checked chan_oh323.c and indeed it only takes one parameter now so I am wondering what was that old parameter for and when did they take it off so I may try pulling that version of asterisk to try with.
The author of that software needs to update because the asterisk API h
Steven Critchfield wrote:
Do we need a presentation on how to behave on the list to avoid getting
flamed by me, or should I just show up with an appropriate LART device
to fix problem people during the normal presentations?
Maybe we should we bring the cattle prod tradition from that other
CO
Daniel Bichara wrote:
Hi All,
Since I updated my * (CVS 2004-03-24), daily, I am getting a strange
message just before a segmentation fault: "Unable to process inband DTMF
on 2 frames".
That message is usually caused by using inband DTMF and using a
compressed codec. All codecs except ulaw an
On a Grandstream ATA and CVS HEAD from last night, and with echo off,
I'm able to receive faxes. With echo on, no go.
HTH,
Ryan
On Apr 15, 2004, at 1:58 PM, Ariel Batista wrote:
Osvaldo Mundim wrote:
Hi,
Does anybody have ATA 188 working with any kind of fax machine? I've
tried many different c
http://bugs.digium.com/bug_view_page.php?bug_id=0001396
Indications for australia. Please confirm if this works for you so we know if this
is something to include in CVS or not.
Thanks, mate :-)
/O
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[EMAIL PROTECTED]
http:
i've been searching the archives but can't find anything substantive on
this. most of the music on hold documentation discusses integrating
with zap hardware, but i am trying to send it across a sip channel.
I have the following in extensions.conf:
exten => 2100,1,Answer
exten => 2100,2,MusicOn
In article <[EMAIL PROTECTED]>,
Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> On Thursday 15 April 2004 03:01, Tony Mountifield wrote:
>
> > This all seems rather cumbersome, and I haven't had the chance to
> > experiment with this feature yet, so the above probably highlights
> > both (a) my lack
Here's the link. It's 7-10 June, in London.
http://pulver.com/europe2004/register.html
Warning: Unlike Santa Clara, they've for some reason decided not to have an
exhibits-only pass available. You have to pay a lot for a full conference
pass to attend. Maybe some feedback would be in order
We also are having randomly dropped calls with our IAX2 connections, we
have tried IAX2 with and without trunking enabled. the phones are snom
200's with SIP and there is an asterisk box at each site so no sip nat
problems.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAI
Walker Haddock wrote:
I just hit google and got this. However, there are 35 hits so take a look.
http://lists.digium.com/pipermail/asterisk-users/2003-October/022796.html
or, more recently:
http://lists.digium.com/pipermail/asterisk-users/2003-February/007855.html
Bottom line, you have to put
Osvaldo Mundim wrote:
> Hi,
>
> Does anybody have ATA 188 working with any kind of fax machine? I've
> tried many different configuration following the Cisco Online Manual
> and I couldn't get this working with Asterisk.
I don't know what the difference is between the 186 and 188 other then the
ex
Hi All,
Since I updated my * (CVS 2004-03-24), daily, I am getting a strange
message just before a segmentation fault: "Unable to process inband DTMF
on 2 frames".
What could it be? Should it cause seg.faults?
Daniel
___
Asterisk-Users mailing list
Can someone direct me to the site for Von Europe? Thanks.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice - 877-203-9308
Fax - 310-943-2606
Dreamtime is yo
Hi,
Does anybody have ATA 188 working with any kind of fax machine? I've
tried many different configuration following the Cisco Online Manual
and I couldn't get this working with Asterisk.
I were trying do change the ATA Connect Mode and Audio Mode reading the
(http://www.cisco.com/en/US/pr
On Thursday 15 April 2004 03:01, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
>
> Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> > If it's a pin-required conference, you will hear the conference
> > number prior to being prompted to enter the associated pin.
> > Obviously, in this case,
I haven't tried breaking up the channels into different groups (mainly because
I haven't had a need to), but the examples I've seen looked more like:
[channels]
signalling=em_w
switchtype=5ess
group=1
context=uti-mainst
channel => 1-3
group=2
context=sales
channel => 4-6
http://bugs.digium.com/bug_view_page.php?bug_id=0001429
We need to architect a general structure for saying numbers in voices.
Say.c is broken as it is now and it needs to be changed.
If we work quickly, this can be sorted out and fixed to 1.1, if not
before that.
There's a number of separate patc
Thanks for the reply.
I didn't include my entire zapata.conf...just the portion that applied
to this call (i.e. group #3)
Please correct me if I have misunderstood how this all works together.
When I see:
-- Executing Dial("SIP/mark-2d08", "Zap/g3/2550559") in new stack
-- Called g3/2550559
directly into voicemail I don't think that's possibile.
but you can fake this function, simply using in the
right way dbput / dbget and if conditions...
Matteo.
Il gio, 2004-04-15 alle 18:45, Iain Stevenson ha scritto:
> Listening to the options on the voicemail system it seems to be missing a
>
Victor Perez wrote:
when trying to build asterisk-oh323 I get the following:
make[1]: Entering directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declara
tions -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/incl
> When we had a MCI ct1, they couldn't send us proper supervised hangup on
> a loopstart encoded DS0. They claimed it to be a problem with the
> software on their switch. Their solution was to switch to groundstart.
> Our end solution was to drop them and switch to Telcove(formerly
> Adelphia) and
On Thu, 2004-04-15 at 12:15, Andrew Kohlsmith wrote:
> > > - Supervision
> > > Loop Start, Ground Start, Reverse Battery, E & M
> > > "DNIS - Dialed Number Identification service"
> > > Not applicable with DS0 Analog Line trunks.
> >
> > You will probably want E&M as you get good answer
On Thu, 2004-04-15 at 18:21, Philipp von Klitzing wrote:
> So to me it looks like IAX2 is involved as well, not just SIP.
Are you sure?
I did some analysis of my traffic. Here is what I found so far:
Only Grandstream phones appear to be affected. All phones affected have
been behind a coned NAT
On Tue, 2004-04-13 at 01:12, James Gardiner wrote:
> Hi all,
> I am not sure if tis is a bug but..
> Was learning about VM etc to see how it all worked, and I noticed the
> following..
>
> In the default install, the VM system leaves 3 different copies of the Voice
> message.
> Size filename
> 13
> PRI section.
> > DS1 Digital Signal: ISDN PRI Only
> > - Primary Rate Interface (ISDN PRI)
> > - If multiple DS1s, Does the customer want
> >"Non-Facility Associated Signaling"? (NFAS) (Y/N)
> I think this is a No, but I am not sure.
I don't think that Zaptel PRI software can currently
> > - Supervision
> > Loop Start, Ground Start, Reverse Battery, E & M
> > "DNIS - Dialed Number Identification service"
> > Not applicable with DS0 Analog Line trunks.
>
> You will probably want E&M as you get good answer detection and hangup
> detection and the easy ability to add DID
It looks like your channel and group statements in the zapata.conf are the
problem. Notice that when it tries to dial out it does so on Zap/6-1. You
have the T-1 defined as 'Span 1,' but you are trying to send the calls to span
6. It ain't gonna work! I don't see anywhere where you've assigned
Listening to the options on the voicemail system it seems to be missing a
feature for users to turn voicemail off completely. This seems a rather
glaring omission. Does the feature of turning off message recording via
the phone exist - or does it need a patch?
Iain
__
On Thu, 2004-04-15 at 09:36, David Stubbs wrote:
> Hi all, Muppet from the UK asking for help
There is 2 sections here. One for channelized T1 and one for PRI. I'll
answer all questions, you choose which one you needed.
Channelized section.
> = Questionnaire Follows =
Tom Green wrote:
Brian,
Encrypted SIP messages can be sent using TLS. However,
I don't think it is realistic to expect everyone
calling you to have a public/private key pair.
SMTP servers that support SMTP-TLS and have valid certs + config do
exactly that already...
--
Best regards,
Duane
http:
Matteo Brancaleoni wrote:
eh, very good idea...
but how about for alaw people?
And E1's and EuroISDN :-)
Any plans to make another conference in EU world?
We'll start with one conference for everyone. As I'm also
based in Europe, having a European followup is an idea
that is within our plans. (Al
Hi!
> > I see this very same effect rather often in the following setup:
> >
> > SIP (GS101) --> *1 --> IAX2 --> *2 --> MGCP (ip10)
> >
> > In fact I think I've seen it also with SIP instead of MGCP at the end.
> > The first client is behind NAT, by the way.
>
> That must be it. I have seen thi
when trying to build asterisk-oh323 I get the following:
make[1]: Entering directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declara
tions -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c
Im interested can you send information?
Kyle
[EMAIL PROTECTED]
- Original Message -
From: "Pertti Pikkarainen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, April 10, 2004 2:26 AM
Subject: Re: [Asterisk-Users] PC based Switchboard application
> We have switchboard applic
Brian,
Encrypted SIP messages can be sent using TLS. However,
I don't think it is realistic to expect everyone
calling you to have a public/private key pair.
Cryptographic solutions have been suggested for email
spams also but they have been found to be ineffective
because of scalability problems.
Voicemail,from wiki pages:
Returns -1 on error or mailbox not found, or if the user hangs up. Otherwise, it
returns 0.
Question is ... how can I trap the error code to divert to a message like "This user
is offline and has no mailbox, goodbye!".
Tnx !
--
Best regards,
Alessio
On Thu, 2004-04-15 at 08:16, Olle E. Johansson wrote:
> We're proud to announce Astricon 2004 - the first Asterisk user's
> and developer's conference!
>
> * Where? Atlanta, USA
> * When? September 22-24, 2004
> We're now in the process of setting up the agenda and are looking for speakers
> and
Hello,
We've made thousands of recording on our Asterisk system, and we had one
last night that ended up having a constant loud hum throughout the entire
recording. None of the other recordings from last night have the hum and the
caller said the conversation had no hum in it. Anyone have any idea
I can transfer a call from my sipura using flash, *98 and , the problem is If
I hangup before the destination extension picks-up, the transfer is lost.
Is there a way to transfer and hangup without having to wait for the destination
extension to pickup?
Regards,
Victor Perez
In a way Europe does make this easy, at the expense of choice.
These questionaires cover the gamut of standard analog lines to
hi-cap service, and ISDN hi-cap.
An E1 is almost (maybe always) ISDN, where a T1 is not, and a PRI
is. Consider it a cultural quirk. Both are 24 channels, delivered
on
On Thu, Apr 15, 2004 at 08:38:32AM -0700, Randy Bush wrote:
> > make install -DNO_IGNORE
>
> h, scary considering i don't need h323. or am i misunderstanding
> something?
NO_IGNORE is going to bypass all of the forbidden lines for all of
the dependencies
--
David W. Chapman Jr.
[EMAIL PRO
On Thursday 15 April 2004 07:45 am, Rich Adamson wrote:
> > > I just went through all this as well. The best thing to do IMHO is to
> > > try to find a way to manually assign IRQ in the BIOS. Also, and this is
> > > what I didn't see at first, some slots SHARE IRQ. Avoid this! If you
> > > are not
Brian Cuthie wrote:
Yeah, for a relatively modern protocol SIP has some surprisingly glaring
omissions, such as:
- certificate based authentication
- encryption
- NAT-awareness
I'd love nothing more to see some decent crypto in the IAX2 protocol, it
already covers the third item on your list
On Thursday 15 April 2004 01:44 am, PTCHEN wrote:
> Hello,
>
> I am an Asterisk beginer and are using it. Now I have 1 question, please !
>
> How many lines of IP phone can Asterisk support, if we use only IP
> interface?
>
>
> Chunghwa Telecom BTA Tech. Lab.
> E-mail:[EMAIL PROTECTED]
That will d
All in all, I was more hoping to get some words of wisdom from the more
worldly Audio Compression experienced people in regard of the question below
about what is the best way to store audio recorded with asterisk.
Ie, to keep the BEST possible quality asterisk can record but still getting
great
Tom Green wrote:
Hi,
Some people have suggested maintaining black lists and
white lists to avoid spammers and allow legitimate
callers into the network. However, the problem with
this method is that the spammer's IP address might
change due to DHCP. Today a spammer might get
aaa.bbb.ccc.ddd and l
> make install -DNO_IGNORE
h, scary considering i don't need h323. or am i misunderstanding
something?
> I'm also working on a freebsd port that uses the cvs version of
> asterisk, let me know if you're interested in taking a look.
o! but i am about to go back on the road. so i d
Hi,
Some people have suggested maintaining black lists and
white lists to avoid spammers and allow legitimate
callers into the network. However, the problem with
this method is that the spammer's IP address might
change due to DHCP. Today a spammer might get
aaa.bbb.ccc.ddd and lets say that I put
> I see this very same effect rather often in the following setup:
>
> SIP (GS101) --> * --> IAX2 --> * --> MGCP (ip10)
>
> In fact I think I've seen it also with SIP instead of MGCP at the end.
> The first client is behind NAT, by the way.
That must be it. I have seen this happening with sip --
Title: Message
I've posted this problem a couple of times before with
little or no response. Basically I have a T100P in my * box.
Incoming calls are working great. However outgoing calls are not working
at all. I've copied a previous post into this message which should have
all the nec
Hi all, Muppet from the UK asking for help
We are just about to have a T1 line installed in our office in Dallas
and "Advantex" the supplier has sent a questionnaire asking a number of
questions. I have put the question area at the bottom of the email, we
will be using Digium's hardware. could any
thanks to those who replied. I have managed to get the functionality I
was looking for working with a series of Macros. However, it doesn't
work as simply as I would like. There are two issues I've run into:
(1)Goto provides no way to pass variables between one context and
another.
(2)I can't f
eh, very good idea...
but how about for alaw people?
Any plans to make another conference in EU world?
Matteo.
P.S. unfortunately I cannot join... too much money for me.
Il gio, 2004-04-15 alle 15:16, Olle E. Johansson ha scritto:
> We're proud to announce Astricon 2004 - the first Asterisk us
On Wed, Apr 14, 2004 at 04:56:46PM -0700, Randy Bush wrote:
> the freebsd port tree version is dead because of the openh323
> issues. before i start hacking, i am hoping someone else has
> a freebsd version that will build on -current. and i do not
> care about h232.
>
> dare i hope?
>
> randy
Hi there,
I am trying to register asterisk to a Lucent's MVAM GK. It is not
registering to the GK with both h323 channels(chan_h323 and oh323).
The problem is that if I set the GK(in asterisk) through the GK's IP, the GK
answers with GCF without it's GK ID, and after that it does not
I'm running the latest version of cvs (not stable), I'm not sure what
the other end is running and if this has been fixed or not yet, however
I was playing round with onhold earlier, the call went to onhold, and
came back from it, then 2 seconds later was hung up unexpectedly, below
is what was
Does anyone know what this means
"Warning [65542]: chan_sip:c:501 retrans_plct: Maximum retries exceeded
on call [EMAIL PROTECTED] for seqno102 (Non-critical Request.
172.16.0.52 is the Asterisk Server
I'm guessing that I have something miss configured just not sure what
it is.
If you need more i
Tobias F. Leucht wrote:
does anyone know if there are channelbanks homologated for the
eruopean/german market, i.e. labeled with the socalled "CE certificate"?
So far, I know that the products from Adtran (TA750 and TSU600) and
Carrieraccess (Adit 600) are not CE-labeled, but I have no alternative
Hello list,
does anyone know if there are channelbanks homologated for the
eruopean/german market, i.e. labeled with the socalled "CE certificate"?
So far, I know that the products from Adtran (TA750 and TSU600) and
Carrieraccess (Adit 600) are not CE-labeled, but I have no alternatives
when it
We're proud to announce Astricon 2004 - the first Asterisk user's
and developer's conference!
* Where? Atlanta, USA
* When? September 22-24, 2004
The conference is arranged in partnership with Digium.inc and the keynote speaker is
Mark Spencer, lead developer of Asterisk - the Open Source PBX. Amo
On Tuesday 13 April 2004 12:12 am, James Gardiner wrote:
> Hi all,
> I am not sure if tis is a bug but..
> Was learning about VM etc to see how it all worked, and I noticed the
> following..
>
> In the default install, the VM system leaves 3 different copies of the
> Voice message.
> Size filename
Make sure you don't have "videosupport=yes" in sip.conf when using
as5300. I found mine doesn't like that much & got that codec error.
Jeremy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Radius
Sent: Thursday, April 15, 2004 2:37 AM
To: [EMAIL PROTECTE
> > I just went through all this as well. The best thing to do IMHO is to
> > try to find a way to manually assign IRQ in the BIOS. Also, and this is
> > what I didn't see at first, some slots SHARE IRQ. Avoid this! If you are
> > not using USB at all, turn it off in BIOS if possible. Tell the BIOS
Hello,
The ONLY issue I have is that I don't get ringing dialback so
calling out gives a silence until the other party picks up
Have you turned on early B3?
S,1,Dial,CAPI/12345678:b${EXTEN}|30 (always early B3)
(plus the recent changes to locks in * required a tweak to the
chan_capi sour
On Monday 12 April 2004 10:19 am, James Moran wrote:
> Is anyone selling asterisks systems??
> Just wanting to know if it's profitable to try and start selling them.
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailm
On Monday 12 April 2004 09:39 am, randulo wrote:
> > Subject: Re: [Asterisk-Users] Booting error - Unable to specify channel
> > 2: No such device
>
> I just went through all this as well. The best thing to do IMHO is to
> try to find a way to manually assign IRQ in the BIOS. Also, and this is
> wh
On Monday 12 April 2004 06:49 am, Andrew Thompson wrote:
> Paul Tyreman wrote:
> >> du -sh /var/spool/asterisk/vm/*
> >>
> >> At the command line, do
> >>
> >> man du
> >>
> >> You will have to know a bit about the operating system, this is not
> >> point and click.
> >>
> >> John Chapman
> >
> >
Hi all,
Below is what I did to run Asterisk in pass-thru
mode:
sip.conf:
[general]
disallow=all
allow=ulaw
canreinvite=yes
For each channel, canreinvite=yes is enabled. No
dial command has 't' option.
However, it seems that Asterisk still stay in the
media path and bridge the 2 end p
It is the same product as listed below, with a different firmware
The firmware does exhibit similar problems
Jason
At 12:42 15/04/2004 +0100, you wrote:
We are currently integration testing the wireless Zyxel Prestige 2000W,
and if all goes well we'll have it for sale in 2 weeks. Has anyone an
On Monday 12 April 2004 04:24 am, [EMAIL PROTECTED] wrote:
Thanks. :) When my *NEW* TDM400P card gets here, I will use
the setup you described and see what happens.
I find your suggestion a bit curious, because the CVS version
I am running (Asterisk CVS-04/10/04-21:44:51) automatically
loads all
On Monday 12 April 2004 01:34 am, Dave Cotton wrote:
> On Mon, 2004-04-12 at 03:39, Anon wrote:
> > On Sunday 11 April 2004 07:18 pm, Todd Lieberman wrote:
> > > do a cat /proc/interupts
> > >
> > > your should see your hardware showup.
> >
> > OK...
> > cat /proc/interrupts
> >CPU0
> >
We are currently integration testing the wireless Zyxel Prestige 2000W,
and if all goes well we'll have it for sale in 2 weeks. Has anyone any
experience of this SIP device and asterisk?
Tan
telappliant.com
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