.101.1: Permission denied,
deleting
WARNING[15516]:pbx_spool.c:388 scan_thread: Failed to scan
service '/var/spool/asterisk/outgoing/smsq.mttx.101.1'
I will try to debug this during the week.
Thanks to you both for your help.
Mick.
Hey Mick,
which version of asterisk are you using ? I've
All working great now!
Cheers.
Mick.
I now have another problem with sending messages back to the phone,
when I run:
smsq -o0198339100 -q101 --mttx-channel sip/phone1 --ud test
You need to run smsq as the Asterisk user, or else the file is created
with permissions
of problems with certain versions of app_sms.
What is working now for me is Asterisk 1.2.9.1, but with
app_sms.c from Asterisk 1.2.7.1.
Get it from here
http://svn.digium.com/view/asterisk/tags/1.2.7.1/apps/app_sms.c?rev=19815
Hope this helps.
Mick.
On Mon, 12 Jun 2006 08:26:38 -0300
Josué Conti
} ${CALLERIDNAME})
exten = s,5,Hangup
I tried with a few different delays.
But I will try with a few more.
Thanks.
Mick.
I tried putting in a delay like you suggested, but it had no effect.
How exactly did you put the delay in? It should be:
Answer
Wait(1) (or .5 or whatever - just play
it on a fixed line and see if I can make it work
there.
Cheers
Mick.
'sa' would appear to be the right option, as Asterisk in your case is
answering the call as the message center (the phone is the 'terminal
equipment')
Would the pap2 be doing anything funny like waiting for fax tones or
something
, but the phone
isn't receiving the messages from * ?
Would anyone have any ideas on how to get this going ?
Thanks
Mick.
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in a delay like you suggested, but it had no effect.
Thanks again for your help.
Mick.
I'm a bit confused about exactly what isn't working... you have given
the asterisk receiving parts of extensions.conf, and say that when you
send a message from the phone to * you get a 'no data' message
if there is currently
support for this in asterisk or not (search the archieves) but its what I do
with our Cisco router and a very neat little windows fax program called
T37FSP from Sandler Consulting. You could prolly use the free version for
testing.
hope this helps,
cheers,
Mick
Nahid Hossain [EMAIL
to the datasheet:
http://www.digi.com/pdf/prd_mca_datafirequad.pdf
Ive only used asterisk with Cisco SIP gateways so Im not sure if this is
enough information.
thanks again for any help,
cheers,
Mick Hastings
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compatible with asterisk?
Using CAPI drivers?
Can you please point me in the right direction for more information for this
card?
Im sure you get the picture here, I really don't know where to start J and
really appreciate your help.
Thanks Mick
can anybody offer any information as to where I should start to look for
more informaion this topic?
I really appreciate the help.
cheers,
Mick Hastings
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.
CroomeAsteriskWinManager
IPSwitchBoard
I use these now and theyre great, they'll work with any(?) SIP phone too,
not just Cisco.
cheers,
Mick.
Nabeel Jafferali [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi.
I have lousy programming abilities, but put this together
here if you fix it.
cheers,
Mick
Eric Rees [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Has anyone seen the error below or knows how to fix this. Every time
this error occurs, I starting getting a 3 second delay on all internal
and external calls and the only why to stop
the help,
Mick
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just fine.
cheers,
Mick
Robert Rozman [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi,
I'd like to setup delayed dial under Asterisk. That means that at the
caller side I set up number *YY and call Asterisk PBX (XXX... is
number of Asterisk PBX, * means pause (2 secs
this resolved.
cheers,
Mick
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OK
I found it in modules.conf. looks like this:
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload = chan_alsa.so
;noload = chan_oss.so
Is this correct?
cheers,
Mick
Jim Kou [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED
a call from the CLI using 'Dial' it just says:
*CLI Dial
No such command 'Dial' (type 'help' for help)
*CLI
the same thing for Answer, Hangup, etc
what have I missed?
cheers,
Mick
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http
,
im no linux guru or c programmer.
I probably missed something in the wiki or something else really basic but
would appreciate any pointers on how to add this patch (or adding patches in
general)
cheers,
Mick
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Hey Olle,
thanks for this info, I have the files but no idea what to do with them, any
pointers would help.
cheers,
Mick
Olle E. Johansson [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Mick Hastings wrote:
Hi Folks,
cheers for all the great info on the list.
I need to create
Hi Folks,
cheers for all the great info on the list.
I need to create a SIP NOTIFY message to reboot my Cisco 7960 phones but I
dont know how.
The admin guide gives an example of the packet (attached), I have tried a
few web searches and found some cool
little programs that generate SIP packets
'Calls through
SIP proxy'
5/ use this profile for your asterisk connection
follow the wiki from there. :)
hope this helps, I edited the Wiki to show these steps in case somebody else
out there has the same problem. I hope this is OK with everybody?
Cheers,
Mick
Norman Zhang [EMAIL PROTECTED] wrote
]
type = friend
host = 192.168.0.200
fromuser = mick
fromdomain = hcjp.com
asterisk -vvvc
Snips
Asterisk Ready.
*CLI Dec 2 16:27:17 WARNING[6754]: chan_sip.c:604 __sip_xmit: sip_xmit of
0x8f2ca6c (len 352) to 192.168.0.200 returned -1: Bad file descriptor
Dec 2 16:27:18 WARNING[6754]: chan_sip.c
Sorry what did you say you had in your hand ???
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, 19 December 2003 2:34 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] after hours
On Thu, 2003-12
When setting
include = daytime|9:00-21:00|mo-fri|*|*
How does this determine what is different between 9 AM and 9 PM
And after hours ???
I want different hours on Saturday and Sunday
And a different welcome message after hours
Any help appreciated
Regards Mick
Stevie
If you do not have any thing intelligent to say
Why waste both your time and ours
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, 19 December 2003 1:40 AM
To: [EMAIL PROTECTED]
Subject: Re
Thanks
Regards Mick
[weekend]
s,1,blahblahweekend etc
[EMAIL PROTECTED] wrote:
Matt
I understand that bit but
How do I express the sound file for after that time period ??
Here is what I need to do
include = daytime|9:00-21:00|mo-fri|*|*
include = weekend|10:00-19:00|sat-sun|*|*
I
Yeah
And give him a gun
And look at what happened at Port Arthur
Regards Mick
Although it's hard to see the original proverb writer saying RTFM
:-)
Matt
Andrew Thompson wrote:
Give a man a fish and he eats for a day. Teach him to fish and he eats
for a lifetime
You can see any thing
Sorry I could not resist
If you need to admin Linux without a monitor
Try webmin
Regards Mick
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on the shelf in a coat closet.
Sadly, there is not enough space
Well I feel you are right there are a few people on this list
That could use a good kick.
Aren't there Andrew
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Thompson
Sent: Friday, 19 December 2003 7:41 AM
To: [EMAIL
Hi all
How can I make * ring one phone then if no answer
Go to a different extension ??
Any help always appreciated
Regards Mick
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Thanks matt
Regards Mick West
NetExpress
Phone 61 08 82420173
Fax 61 08 82425099
[EMAIL PROTECTED]
Disclaimer:
Confidentiality:
This message contains privileged and/or confidential information
intended only for the use of the addressee named above.
If you are not the intended recipient
Hi again
How do I change the message played on initial pickup for after hours ??
Thanks in advance
Regards Mick
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What are you after
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of vocalvoip
Sent: Thursday, 18 December 2003 2:15 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] voip equipment in australia
Hi
Anyone in australia know of any good
Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of matt
Sent: Thursday, 18 December 2003 2:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] another
[EMAIL PROTECTED] wrote:
Hi again
How do I change the message played on initial pickup
Thanks for that
One question how do I stop * from picking up that line
But still allow it to dial
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Wood
Sent: Thursday, 4 December 2003 11:25 AM
To: [EMAIL PROTECTED]
Subject
Hi
I have a second line that we use for a fax server
Since we are luck to get 2 faxes a week
I want to use this line as a dial out line for *
But still need to be able to send and receive faxes on it
Has anyone got any ideas how I could accomplish this ??
Regards Mick
Thanks for that
One question how do I stop * from picking up that line
But still allow it to dial
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Wood
Sent: Thursday, 4 December 2003 11:25 AM
To: [EMAIL PROTECTED]
Subject
That is their new price
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh
Roberson
Sent: Monday, 1 December 2003 7:00 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] cisco 7960 power suplies?
Also, I see them on eBay all
Cisco
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lists
Sent: Monday, 1 December 2003 10:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco 7960 power suplies?
Does anyone know where to get cisco 7960 power suplies? What
Surely dick smiths have something
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Thursday, 27 November 2003 4:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] door phone
[EMAIL PROTECTED] wrote:
Hi
Or it could be worse still
They could be you
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, 25 November 2003 6:39 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] unsubscribe
Why do we
Where can I buy the Wifi600 phones ??
Or does anyone know of other Wifi SIP phones ??
Any help would be appreciated
Regards Mick
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Thanks Miguel
But from what I read these phones are not available for shipment until
January 2004
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miguel
Cavazos
Sent: Wednesday, 19 November 2003 2:38 AM
To: [EMAIL PROTECTED]
Subject: Re
yes
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Romanov
Sent: Wednesday, 19 November 2003 9:40 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] FXO Card/Interface for Australia
Hi Guys,
Pardon me for the might be stupid
WARNING[1242952640]: File app_dial.c, Line 318 (wait_for_answer): Unable
to forward voice
This is what I get
And a crash
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, 17 November 2003 5:14 PM
Don't sound bad do they
Except their ship date is mid January 2004
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain
Stevenson
Sent: Monday, 17 November 2003 10:34 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] wireless
I think if he is unable to understand what we have typed
He could be a bit more polite in asking us to explain it to him
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, 18 November 2003 10:11 AM
To: [EMAIL
Has anyone got a mobile wireless phone working with * yet
Is it possible to use the Cisco 7920 with skinny
Regards Mick West
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No
No sip image for it yet
Also is there any way I can change messages and extensions depending on
local time ??
Also is there a way to transfer the call over PSTN if the local
extension is not answered.
Eg to a normal gsm mobile ??
Regards Mick West
NetExpress
Phone 61 08 82420173
Fax
Does anyone know how to make
Calls auto transfer to a mobile if no one answers ??
Regards Mick West
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Has anyone tried to use a Cisco 7920 with *
Regards Mick West
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When making a call from my 7940G out over our Voicetronix openline4 to
PSTN
Our volume is low
Any ideas ???
Regards Mick West
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Tis cool I fixed that 6 hours ago
Thanx for the help any way
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, 7 November 2003 10:17 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voicemail
Sean
Thanks for the response
The call was external over PSTN
All I did was enable max time for voicemail.
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean P.
Robertson
Sent: Friday, 7 November 2003 10:19 AM
To: [EMAIL PROTECTED
There were some fixes posted the other day.
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Bagyenda
Sent: Tuesday, 28 October 2003 3:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: Voicetronix OpenLine4
Hi Jorge
If I receive a call it is fine
But if I make a call the voice level is not so good and the echo is
shocking.
Any ideas would be appreciated
Regards Mick
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expecting
625
Any ideas
Regards Mick West
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???
Regards Mick
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No music on hold when activated from cisco phone
Console debug
EBUG[1116941120]: File res_musiconhold.c, Line 280 (monmp3thread): Read
372 byt
es of audio while expecting 1600
Any ideas would be appreciated
Regards Mick
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onestate[chdev] == 0' failed.
Aborted
This only happens if in
Extensions.conf I place a r in the extion line eg.
exten = 1004,2,Dial(sip/[EMAIL PROTECTED],20,r)
If I leave the r out the phone rings once and then hangs up.
Regards Mick
--
Ben Kramer [EMAIL PROTECTED
when I put a station on hold I receive this message
res_musiconhold.c, Line 280 (monmp3thread): Read 372 bytes of audio
while expecting 1600
Regards Mick
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when I dial out from my Cisco phone I get this error
File channel.c, Line 2258 (ast_channel_bridge): Didn't get a frame from
channel: SIP/210.9.49.216-c26e
Regards Mick
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You can find the defaults at Cisco
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, 18 October 2003 12:13 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] chan_skinny XML Files for 7920
Hi,
I
I am still having trouble changing the sound files.
I can take a wave file out of another program and set it in the folder
and it will work
If I record a wave file in Windoze
No go
Am I missing some thing ???
Thanks for the help
Regards Mick
tar
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Pounder
Sent: Thursday, 16 October 2003 8:16 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] dialling out
something in the dialplan has too course a filter and is matching
Woody
It may have been early but it was not that early.
I was not calling the same number that I was dialling from.
This system is not in production so it is on a spare line
And I was calling our main number
So there is n issue there somewhere ???
Regards Mick
-Original Message
When trying to dial out
982420173 our main number
I get the engaged signal before I finish entering the phone number
Any ideas
Regards Mick
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I am still having trouble changing the sound files.
I can take a wave file out of another program and set it in the folder
and it will work
If I record a wave file in Windoze
No go
Am I missing some thing ???
Thanks for the help
Regards Mick
No I am using a Cisco 7940
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen R.
Besch
Sent: Tuesday, 14 October 2003 11:13 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] dialling out
Mick,
If you're using the Grandstream
) exited non-zero on 'vpb/1-3'
-- hangup on vpb (vpb/1-3)
-- Hungup on vpb/1-3 complete
-- Event [12=[02] Loop Drop
And it hangs up the line any ideas ???
Regards Mick
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how do you go about replacing the sound files in *
with your own ??
Regards Mick
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tar
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Monday, 13 October 2003 9:54 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] replacing sound files
[EMAIL PROTECTED] wrote:
how do you go about replacing
)
== Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3'
-- hangup on vpb (vpb/1-3)
-- Hungup on vpb/1-3 complete
and the line disconects
Regards Mick
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anyone out there got * to work with
a voicetronix openline4 card ???
Regards Mick
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Andrew
I am having trouble with
Sound ( only if you dialling from outside )
Cisco phone can not dial out
If I phone in and select extension number of Cisco phone
* dies
Any ideas ???
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Title: Message
We
need an consultant
to
set-up a single line server as demo for our customers
needs
to run
voicetronix openline4 card
Cisco
7940 phones
please
email me off line
Regards
Mick
You need to remember you are looking at it different to what they would.
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester
Sent: Saturday, 11 October 2003 1:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Marketing Digium
That's OK
Easy to get involved and lose sight of the reason these customers
A couple of companies that I found to approach were
Companies with multiple offices
With lots of calls between them and spending thousands of dollars a
month.
Just food for thought
Regards Mick
-Original
I have a new 7940
I have set-up the network
And tried to tftp SIP ver. 2.1
And ever time it boots and starts the tftp download the 7940 reboots
Any input welcome
Regards Mick
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All I get is
Version Error
When trying to tftp
Any ideas ???
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Thursday, 9 October 2003 5:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 7940/60 TFTP Problem
Well I eventually got the 7940 loaded
Now does anyone have quick fix to get it to work with asterisk
Tar in advance
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Thursday, 9 October 2003 9:41 PM
To: [EMAIL
)
Now my issue is I can call in but can not get to the extension ( Cisco )
And from the Cisco phone I can not call out ( pstn )
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Babak Pasdar
Sent: Thursday, 9 October 2003 10:57 PM
I set mine up like this
exten = 1234,2,Dial(sip/[EMAIL PROTECTED],20,r)
And everytime it rings I get
exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp
line:872
And * falls over
This is with a voicetronix openline4 card
Any ideas ???
Regards Mick
-Original Message
Title: Message
has
anyone got a voicetronics openline4 card working ??
If so
do you have any notes etc.
thanks
in advance.
Regards Mick
Title: Message
got
the voicetronics openline4 card sort of working
I keep
getting this error
-- Event
[7=[03] Record fifo overflow] on vpb/1-4
and
the auto attendant is not clear.
thanks
in advance
Regards Mick
Title: Message
I am
trying to set-up asterisk with voicetronix openline4 card
does
anyone one know if this will work or have and ideas on how to make it
work
I have
it in but auto assistant is choppy
any
help would be appreciated
Regards
Mick
I am trying to set-up asterisk with voicetronix openline4 card
does anyone one know if this will work or have and ideas on how to make
it work
I have it in but auto assistant is choppy
any help would be appreciated
Regards Mick
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