On Monday 08 September 2003 11:22 pm, Timothy Soos wrote:
Hello All,
Is there any way to [monitor and record] a channel that is already active
(meaning 2 callers are already connected)?
I read the documentation for the Application commands Monitor and
StopMonitor, and I get the impression
See this big fat newbie over view
http://www.vlug.org/vlug/meetings/presentations/VLUG-Telephony.pdf
-Original Message-
From: Kevin K [EMAIL PROTECTED]
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Date: September 8, 2003 9:01 PM
Subject: [Asterisk-Users] Asterisk as a GW or PBX?
Hi all,
I have just started to play with callgroups and pickupgroups..
I updates my * from CVS this morning (about 15 mins ago)..
I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in
sip.conf..
I place a call from phoneA to phoneB, then I go to phoneC and dial *8# ,
I think *8# is only in the Zap channel driver. There was talk on the list
recently (past day or two) about moving these hard coded codes (like *70 for
call waiting, *67 to block caller ID etc) out of the Zap channel driver and
in to the dialplan so they could be used on SIP and provide some
Hello members,
Does anyone has experience with SNOM 4S Soft Switch. Is there any
comparison matrix available w.r.t to Asterisk.
Thanks,
Tarun
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Rediff Matchmaker strikes another interesting match
Hi,
Has anyone had a play with the Zultys 4x4 IP phone, if so, did it work fine?
Reference: http://www.zultys.com/summary_ZIP4x4.htm
Thanks,
Andrew Griffiths
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What's the secret to getting sound through Xlite? The SIP messages all look
OK to me, but the sound isn't coming through.
It was trying to use GSM, so I searched the archive and tried:
disallow=gsm
allow=ulaw
Now it says that it's using ULAW but I still get no sound in either
direction.
Phil
At 01:02 -0700 9/9/03, [EMAIL PROTECTED] wrote:
Hi,
Has anyone had a play with the Zultys 4x4 IP phone, if so, did it work fine?
Reference: http://www.zultys.com/summary_ZIP4x4.htm
not yet - I am ordering a couple of eval units later today. UK units if
that makes a difference. I would also be
OK you are correct..
*8 picks up the call..I wonder why *8# does not work??
I also had the same problem that the phone that I collected the call from did not stop ringing..
I have problems with this as well ( similar config ). My CVS is 10 days
old.
I can get the call picked up with *8
hello!
Is maybe anyway that asterisk supports this incomplete address
response SIP 484 (early dial) ?
I think it would be nice feature for dial with hard sip phones .. now
must wait with all sip phones ~ 4 seconds or press # ,
but when your dial frequency is high is this somehow
On Tue, Sep 09, 2003 at 08:23:49AM +, WipeOut . wrote:
OK you are correct..
*8 picks up the call..I wonder why *8# does not work??
I also had the same problem that the phone that I collected the call
from did not stop ringing..
I have the same problem. Mark Spencer is working on the
Has anyone put together a template if you will can be used to generate
feature codes for Asterisk. Even if some of the feature codes are
dialplan specific, if there was a template, it could be changed
accordingly. Besides the feature codes, there could also be something
for voicemail, etc.
Just a
hi!
Is it possible to get working message button on grandstream-budge tone
phone ?
For call to VM and also to signaling messages in VM ? or at least any of
this two.
tnx,
Thomas
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Hello,
What configuration should I use for this (I use sip phones)?
Best regards,
Daniel
WipeOut . a crit:
OK you are correct..
*8 picks up the call..I wonder why *8# does not work??
I also had the same problem that the phone that I collected the call from did not stop ringing..
snip
In this configuration I had to forward the iax port on both NAT boxes to the
* box.
192.168.0.100 -NAT1- publicIP - publicIP - NAT2 192.168.0.100
It wasn't sufficient to register from one * box to the other to get audio
working because the source port was different. * was version 0.4.0 and
Tim,
Using Monitor you can record an already active call; give it a try!
Dave
=
David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales
ToadNet - Want to go fast?410-544-1329 FAX
570
Yes. They are on the same subnet.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: 09 September 2003 11:12
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Xlite = no sound
On Tue, Sep 09, 2003 at 09:15:32AM +0100, Skuse, Phil wrote:
What's the secret to
On Tue, Sep 09, 2003 at 11:41:17AM +0100, Skuse, Phil wrote:
Yes. They are on the same subnet.
I solved my sound problem with X-lite by using the latest
CVS version and compiling that. I had been using the
stable and unstable versions out of Debian.
Where did you get access to X-Ten.com's CVS server?
I didn't know they had the source code for x-lite available..
Later.
On Tue, Sep 09, 2003 at 11:41:17AM +0100, Skuse, Phil wrote:
Yes. They are on the same subnet.
I solved my sound problem with X-lite by using the latest
CVS version
Pick up an X100P - http://www.digium.com
Jeremy McNamara
Angel Gabriel wrote:
I want to play around with asterisk... I have a machine with a modem in
it, and I want to learn configuration and how to use it. But someone
told me, that this system doesn't work woth modems if this is the
case,
- Original Message -
From: Paul Crick [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 09, 2003 10:16 AM
Subject: RE: [Asterisk-Users] Callgroup, Pickupgroup and SIP
I think *8# is only in the Zap channel driver. There was talk on the list
recently (past day or two)
Hello!
I have a staff member abroad and need to provide him with the ability to
make local calls. The features I need are:
* Possibillity to make calls at local (Icelandic) charges from Ireland
office.
* Possibillity to call the local Icelandic number and reach the Ireland
office.
I'm also
It's exactly what I have done, I have this log message:
NOTICE[114696]: File chan_sip.c, Line 4870 (handle_request): Nothing to
pick up
WARNING[114696]: File chan_sip.c, Line 2220 (__transmit_response):
Unable to determine sequence number from ''
From 276 I dial 326 and trie to pickup using
When last did you update from the CVS??
But it curently doesn't work anyway so just hang in there and hopefully it will be fixed soon..
It's exactly what I have done, I have this log message:
NOTICE[114696]: File chan_sip.c, Line 4870 (handle_request): Nothing to
pick up
WARNING[114696]: File
On Tue, 2003-09-09 at 00:33, Troy Settle wrote:
Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too
expensive.
Can anyone recommend a decent channelbank that won't break the bank?
Since those parts are not commodity yet, you get what you pay for.
Zhones are a pain to
WipeOut . a crit:
When last did you update from the CVS??
It's from 2003-09-04 CVS
But it curently doesn't work anyway so just hang in there and hopefully it will be fixed soon..
It's exactly what I have done, I have this log message:
NOTICE[114696]: File
When a Snom 200 (v2.1l) calls a C7960 (v4.4), both using g711u as default,
the conversation is extremely noisy from the Snom to the Cisco, but clear
in the reverse direction. Using a sniffer, I see packets from the Snom to
the Cisco of 87 bytes and Cisco to Snom of 214 bytes. Asterisk is CVS
Hi,
I have a delay betweentwo H323.
Netmeeting1 - |
|
|gnuGK
| --- [asterisk-oh323] | Asterisk |
Netmeeting2 --|
|
Netmmeting 1 call Netmeeting 2. When Netmmeting 1
speak Netmeeting 2 receive the voice without delay. But in the other way I have
3
Hi all.
On 2 october will start SMAU, here in Italy , in Milano.
SMAU is the biggest IT (and computer related stuff) expo event
that we have in italy.
I'll be @ SMAU from 2/10 to 6/10 , in the opensource area,
where my company will promote asterisk digium hardware.
If anyone will attend the
Troy Settle wrote:
Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too
expensive.
Can anyone recommend a decent channelbank that won't break the bank?
TIA,
I've bought several vintage Premisys CB's on ebay for $75 to $150 /ea.
They work fine with my T100p's. This is not
Hi all,
is it possible to disable RTP routing through asterisk? RTP routing is a
very nice feature but, I think its important also to disable it in some
cases (e. g. in a LAN).
Do you have any suggestion?
Andrea
Rattana BIV wrote:
Hi,
I have a delay between two H323.
Netmeeting1
Add dial plan entries like this in extensions.conf
[trunks-ld]
; long distance
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],20,Tr)
exten = _1NXXNXX,2,Congestion
add a sip entry like this in sip.conf
[mt-1204]
type=peer
host=172.20.16.7
mask=255.255.255.255
dtmfmode=inband
This one seems less expensive. Does anyone have any experience with Rhino?
http://www.channelbanks.com/
We don't have one yet, but probably will soon.
Regards,
Neal
Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too
expensive.
Can anyone recommend a decent channelbank that
I have the same problem with Cisco 7960s. I have found however that if
you do the *8 from a Zap channel, then the called phone stops ringing.
But if I do *8 from a SIP device, the called phone continues to ring and
ring for at least a minute.
I think Mark is working on fixing this (or at least
FWIW, I also bought a rack of Tellabs echo cancellers for $50 and have
one between the cb and the T100p. A little overkill, but very effective.
Hey hey..
What are the model number of these Tellabls echo Cancellers
do you then turn off all asterisk echo Cancellation in zapata
Do you tuly have 0
yes, i had 'callprogress=yes', and i commented it.
now the time out is working.
Thank you very much
by disabling callprogress in an analog environment, does it affet the
call disconnection?
IT does but you should only use it for analog channels ... and propably
only FXOs.
Martin
Surajee
For anyone who is interested it appears that the nes Snom firmware now gets the phone
to poll for updates approx every hour.. I was wondering why my ISDN line was being
activated all through the night..
I see there is a setting now to control updates so I am going to set that to Never
update,
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Hash: SHA1
Hi,
How do I optionally hide the caller id on outgoing calls on chan_zap? Ie.
calling h323 - asterisk - chan_zap - isdn provider.
Using setcallerid() to clear the callerid won't work since my provider
requires a callerid. AFAIK one has to send
I had posted earlier asking about a Snom200 communicating with a C7960
and lots of noise in one direction. Turned out the problem was created
by me removing the allow=all statement in sip.conf. Someone had suggested
that statement is no longer needed, and using allow=ulaw, etc, had an issue
where
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Tuesday, September 09, 2003 1:26 PM
To: Asterisk-users-list
Subject: [Asterisk-Users] Has the allow=all function
changed in
At 11:26 AM 9/9/2003 -0600, you wrote:
What is the general consciences for the allow=all statement? Should it be
used, should it be specific towards those codecs supported, or removed?
My understanding is that you MUST have at least one allow and one deny, or
none at all. Just having one or the
All,
I've been doing a lot of work on the Queue application and have some ideas
I wanted to get some feedback on. Many people, myself included, have
stated their desire for more/better queue statistics. We ought to be able
to bring things up to a standard comparable to the established ACD
David,
Could you elaborate on what hardware you're using for this?
Also, speaking of faxes and *, I know that you can't fax over IAX(2),
but it seems that faxing over SIP (ala ATA-186) works fine? Would it be
possible to set up the fax extension on one * box that can then use SIP
to get the
On Tuesday 09 September 2003 07:17, Maron Kristófersson wrote:
Hello!
I have a staff member abroad and need to provide him with the ability
to make local calls. The features I need are:
* Possibillity to make calls at local (Icelandic) charges from
Ireland office.
Sure. Just send the IAX
You betcha!! We use it at work and we actually have non-html people
contributing and fixing typos and errors..its easy to set up and easy
to use.
Leif Madsen wrote:
Hm
http://www.interactivetools.com/products/docbuilder/
This looks kind of what I want, but I am looking for a
You could do it with a couple of soft phones if you have enough PCs to go
around - I did this a little bit, just to play. Then I forked out for a
Cisco 7940 and ATA-186 off eBay so I had some real phones to play with..
then I forked out for the Dev Kit Lite (see - it gets addictive, this
Asterisk
FYI I asked them:
Your website talks about configuring the Rhino channel bank
as 24xFXS.
Is it possible to mix FXO and FXS modules? What affect does
that have on pricing?
They replied:
We will have FXO and the ability to mix both FXS FXO within 60-90
days. Our RD department is testing this
I have an ISDN TA that has 2 POTS interfases (FXS), can these be used
with asterisk?
Thanks in advance
Robb
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On Mon, 8 Sep 2003, Jim Mercer wrote:
Can we bribe you? :)
sure, pay my rent for 3 months and give me a 50 plasma TV to play in the
background.
Is that all? That sounds rather cheap, compared to the things direction
that I'd have to go if I wanted to stick to the cisci CM route, with
Hello,
this is my first post to the list. I have to say that i am really
impressed with *. Every day i find a new great feature to play
around with. Also the SIP support has come a long way in just half a
year. Just wanted to thank the programmers for this great software.
I have one problem
Know bug
http://bugs.digium.com/bug_view_page.php?bug_id=116
Pertti Pikkarainen wrote:
I have problems with this as well ( similar config ). My CVS is 10
days old.
I can get the call picked up with *8 ( *8# does not work ) but
the phone B never stops ringing.
B rings forever. I'm
know bug http://bugs.digium.com/bug_view_page.php?bug_id=116
WipeOut . wrote:
I have just started to play with callgroups and pickupgroups..
I updates my * from CVS this morning (about 15 mins ago)..
I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in
Hi David, group..
I definitely agree that some kind of external statistics would be of great
benefit. I implemented some event logging on an IVR system I worked on
recently, where events were stored in a table related back to the unique ID
in the CDR record (so one CDR, many events) and it worked
The hardware we are using for the asterisk server is a 2U server with dual
Xeon procs. It has a 4-port T1 card in it. Two ports are used for voice T1s.
Port #3 connects to an Adit 600 channel bank with 8 FXO ports and 16 FXS
ports. Our hylafax modems are plugged into the fxs ports. The modems we
On Tue, Sep 09, 2003 at 11:04:34AM +, WipeOut . wrote:
Where did you get access to X-Ten.com's CVS server?
I didn't know they had the source code for x-lite available..
Sorry, I should have been more clear - I used the latest
version of Asterisk via CVS.
As a first step towards getting real queue statistics going (a project
arguably adjunct to Asterisk itself), I propose that the Queue app be
modified to produce its own Queue Log, which could be stored either as
text or in a SQL table. For simplicity I would probably just implement a
MySQL
I wonder if your company took your name off of the support contact page
because you do not know how to properly use your email software yet.
Can you justify your abuse of the reply button where you only changed
the subject line and didn't even remove the non relevant message below?
Stupid,
I thought I saw a thread that said that a pseudo modem driver was built, but
only for h.323.
Could it also work for SIP?
Are there any details?
Thanks
Lee Goodman
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ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt
The text you want is on pages 182 - 193.
/a
-Original Message-
From: Steven J. Sobol [mailto:[EMAIL PROTECTED]
Sent: Monday, September 08, 2003 12:35
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP Status Codes
Can anyone
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the same codec.
On Tue, 2003-09-09 at
At 02:38 PM 9/9/2003 -0500, you wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the
I've played with LED readerboad signs before and they're pretty easy to
program up - Does anyone else on the list have any interest in making these
ACD stats available in this way?
I have done some trival work with matrix orbital lcd to show some stats
counts, calls parked etc
Just find lcd a bit
These are 2551 32ms echo cans. Each does 1-T1 and sits between the T100P
and the cb.
Yes, the zapata echo canceller is turned off.
And, the echo is gone. I can't even detect the training time.
CL
Software is obviously a much more tidy approach, but they were cheap,
echo was a problem
Heheh..lets try that again...we use TWiki http://twiki.org at work..and
I'm in the process of setting it up at home..its a GREAT way to
document anything. It starts as chaos and then eventually builds up a
well defined structure as you go along.
Chris
Chris Hirsch wrote:
You betcha!!
Thank you. I saw references to these two settings in various places but did
not see how both were used together.
I appreciate it.
Sean
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 09, 2003 3:38 PM
Subject: Re: [Asterisk-Users]
I heard about this a while ago too. How come I didn't hear anything about
it from asterisk-announce? (at least I don't recall receiving any emails
about it.
Also, is there any plans in the future to create a stable and development
branches of code? Upgrading to the lastest CVS version may be
On Tue, 9 Sep 2003, Eric Wieling wrote:
Transcoding would be required for access to ANY of the asterisk
sound files, voicemail and PSTN via Zap interfaces.
If you are using G711 ulaw from the SIP phones, and that is what
you are getting from the T1 PSTN link, would * have to transcode
that?
On Tue, 2003-09-09 at 15:23, Brian Jones wrote:
I heard about this a while ago too. How come I didn't hear anything about
it from asterisk-announce? (at least I don't recall receiving any emails
about it.
Also, is there any plans in the future to create a stable and development
branches
I have done some trival work with matrix orbital lcd
to show some stats counts, calls parked etc Just find
lcd a bit small do you have lead on bigger LED signs
that you have used b4 ??
I've used a Beta-Brite sign which is pretty similar to a ProLite in
functionality, just made by a different
On Tue, 9 Sep 2003, Paul Crick wrote:
I have done some trival work with matrix orbital lcd
to show some stats counts, calls parked etc Just find
lcd a bit small do you have lead on bigger LED signs
that you have used b4 ??
I've used a Beta-Brite sign which is pretty similar to a ProLite
Hi,
We are installing a new Asterisk PBX system and need to find a VoIP carrier
that will handle all our long-distance services and tie in directly with the
Asterisk server. Any recommendations? I've called NuFone and was not impressed
by their voicemail answering system (choppy) and was unable
Any suggestions? The Aastra 480 390 seem popular along
with the CybioLink.
Does anyone use these phones (or others)?
Are they compatible with atsterisk's ADSI?
If so, how are people programming these phones?
Searched thru archives, lots of previous talk but
no soild info. I'd like to get a couple
It would have to do some kind of trascoding, but it's a non-issue since
G729 is not involved and the CPU overhead is minimal.
On Tue, 2003-09-09 at 15:26, Mike Ciholas wrote:
On Tue, 9 Sep 2003, Eric Wieling wrote:
Transcoding would be required for access to ANY of the asterisk
sound
Is there a basic web interface to the console to the asterisk
system like webmin?
There was a ton of discussion on the Aastra's. Please check the list
archives:
http://lists.digium.com
-Original Message-
From: Ken Godee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 09, 2003 4:54 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ADSI phones?
Any
Hello,
this is my first post to the list. I have to say that i am really
impressed with *. Every day i find a new great feature to play
around with. Also the SIP support has come a long way in just half a
year. Just wanted to thank the programmers for this great software.
I Agree!!
I
I have asked this question in the start, but then realized that this
qustion does not generate any response at all. Going through the archive I
find out that lot of people have claimed to have made web gui, but when I
see them, I stick to vi.
It should be very easy to do, generate config files
On Tue, 2003-09-09 at 11:41, Hielke Christian Braun wrote:
I have one problem with the BudgeTone phones and early dial. When i
dial a long external number with 9+, * starts to dial to early with
just a few digits. The outgoing call is placed through the SIP provider
Nikotel. Is there some
I've got an Aastra 390 (from Telus in BC, Canada) and have had a successful
ADSI script download. The Comedian Mail graphical interface works on it too.
I had a couple of glitches where voice prompts were being played but the
phone wasn't responsive but these aren't repeatable. I'd grab a phone,
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On Tuesday 09 September 2003 17:54, Tais M. Hansen wrote:
How do I optionally hide the caller id on outgoing calls on chan_zap? Ie.
calling h323 - asterisk - chan_zap - isdn provider.
Problem solved. I made app_dial.c take an option to change
It should be very easy to do, generate config files (or meta
config files) from web gui.
Do you mean like storing config information in a separate database, then
generating the plaintext config files everytime there's a change? This is
very similar to what a lot of the web hosting control panels
We have some regular POTS phones connected to our incoming line as well
as the machine that runs Asterisk. Sometimes during an outgoing call
from the POTS phone, the Asterisk will pick up also, and play its menu.
The FXO card is set to fxs_ks signalling; I'm told this might be the
culprit but
On Tuesday 09 September 2003 16:46, Steve Meyers wrote:
On Tue, 2003-09-09 at 11:41, Hielke Christian Braun wrote:
I have one problem with the BudgeTone phones and early dial. When i
dial a long external number with 9+, * starts to dial to early with
just a few digits. The outgoing call is
The only way I was able to solve my delay issue with Chan_oh323 was to
switch to Chan_h323.
Chan_oh323 caused a similar 3 -4 sec delay on one way of the conversation.
Checking the CPU stats on asterisk during the call - confirms that the RTP
stream was somehow routing through asterisk - not
Anybody have any experience pushing XML display data to a Cisco 7960? I
am pulling data from a webserver now, but I have stuff, like queue status
info, I'd love to be able to push at particular phones.
A pick up the damn phone push message would be cool too. :)
Dave
Hi
I really need caller id to work in the UK, I understand that the X100p
uses a US chipset,two questions
1) is that a product that converts UK to US caller id in line
or
2) would it be possible to have modem that supports CID in parallel
with the line and the x100p.The modem reads the line
On Tue, 2003-09-09 at 16:03, Tilghman Lesher wrote:
Why not just use DISA:
exten = 9,1,DISA(no-password|outgoing)
Because I didn't know about it. :) I'll try it out.
Steve
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yes, I agree with you.
I verify with a sniffer and asterisk manages RTP flows. The problem is asterisk
decode and then code again RTP flows. This function requires 5-7% CPU On my
test-box (Linux rh 7.3 on P3 600 GHz). This solution don't scale without dedicated
HW, I think!
Another problem is
Hi all,
I'm interested in using asterisk WITHOUT codec support: I work in a LAN, with no
bandwidth, delay, ... problems; I use a Cisco GW as PSTN interface and when I
use asterisk the overall delay is to high and the quality drops.
In particular, I'm interested in using asterisk as h323 to sip
Yeah, the only reason I ask is that pushing a URL is apparently possible
with the Call Manager firmware. See Cisco::IPPhone perl module. Was
hoping similar concept was possible for the SIP firmware... so if anybody
knows...
Dave
On Tue, 9 Sep 2003, denon wrote:
I'd hate to see development efforts get split up, and more time spent on
porting/etc efforts, detracting from primary development. If it's now
slowing down new development, it's always a step behind while someone
patches up the current builds.
Once the code
Travis Johnson wrote:
I've called NuFone and was not impressed by their voicemail answering system (choppy) and was unable to even leave a message before the phone call was disconnected (in the middle of the
recording).
So your going to judge our system by making one phone call into my home
The show modules command line suggests that is Not loaded. Should if be, and
if so, where do I do that?
Rich
Do you have loaded ztdummy as module?
-Mensaje original-
Trying to test the music on hold function and can't seem to get it to work.
If anyone has
At 07:57 PM 9/9/2003 -0400, you wrote:
Travis Johnson wrote:
I've called NuFone and was not impressed by their voicemail answering
system (choppy) and was unable to even leave a message before the phone
call was disconnected (in the middle of the
recording).
So your going to judge our system
I agree 100%. Sometimes code works on one system
because of a quirk. Building and testing on
multiple systems and debugging the autoconf scripts
has a way of making the code mature and robust. I
keep and old DEC Alpha (a 64 bit machine) and SPARC
(big endian byte order) and run Solaris and
If I'm not mistaken, ztdummy makes work moh and meetme...
Do lsmod from sh. If you don't see ztdummy, download zaptel, and modify
Makefile. Line 89 must be:
MODULES=zaptel.o tor2.o torisa.o wcusb.o wcfxo.o wcfxs.o \
ztdynamic.o ztd-eth.o wct1xxp.o wct4xxp.o ztdummy.o
(Remove # preceding
The source code for your kernel is not the same as the kernel actually
running on the machine.
On Tue, 2003-09-09 at 20:38, Rich Adamson wrote:
After following your suggestions, I get
[EMAIL PROTECTED] zaptel]# /sbin/modprobe ztdummy
/lib/modules/2.4.20-18.9/misc/zaptel.o: kernel-module
another. I mean, by the time you strip down the linux install to make it
rock-stable, it's basically a kernel, compiler, and a handful of
dependances. Does it really matter what OS it runs on at this point, as
long as it's a robust kernel?
At least in my case, I /can/ strip down a FreeBSD
Sometimes it happened that error to me. Reinitiating Linux and proving again
(modprobe/insmod), the error disappeared.
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 09, 2003 9:48 PM
Subject: Re: [Asterisk-Users] help on MOH
The source code for your kernel is not the same as the kernel actually
running on the machine.
I don't see that. Yes, sure.
- Original Message -
From: CW_ASN [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 09, 2003 10:00 PM
Subject: Re: [Asterisk-Users] help on MOH
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