陈帆 wrote:
steven,,
u not understand the situation,,,
there only have 30 channel for voice,, the other channel resiverd for
control channel..
good luck for you.
You seem to have a problem counting. 1 to 15, plus 17 to 31 == 30 voice
channels.
Steve
On 5/10/06, *Steve Underwood*
Lee Howard wrote:
Olivier Krief wrote:
For example, it seems that Brother 8360P uses Super G3 mode.
Is there a fax-modem offering such capability so that I could easily
check if I still cannot hangup when I enable or disable Super G3 mode ?
MultiTech 5634-series and MainPine RockForce
Hi,
I have a queue that plays music when a call comes in. To be able to do
that I need to Answer() the call first. After a timeout in this scenario
the call should be transfered to an extension using a GoTo statement to
the extensions context. The problem is that as soon as asterisk Answers
Which Actions and events to the read/write options in manager.conf give access
to, ie the options below.
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Are they documented somewhere?
/Obelix
___
I recently installed one of these plugged into an ATA with a dialplan that calls a predetermined number when the line is picked up. The sound guys have only to push one button and we get audio into the pbx. Works flawlessly. We did have to boost the input level more that I think we should have
Hi,Thanks to Cory's and HTH's help (spell ? black magic ?), I could at last unlock Network Configuration's menu.To unlock, I pressed Settings key then a combination of # and * keys from dialpad and it worked.
I really thought I've tried every possible combination before writing to this list but
Urban wrote:
May 21 11:03:10 WARNING[12188]: channel.c:2049 ast_indicate: Unable to
handle indication 3 for 'SIP/XX-c52d'
You don't have a /etc/asterisk/indications.conf
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
Hi,I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ?From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2
, I got the following:
Copy the desired binary image from Cisco.com to the root directory
of the TFTP server.
Hi,
I have a TDM400P and it works,
Send us the output of your dmesg command.
Regards.
--- M.Hockings [EMAIL PROTECTED] a écrit :
In my attempt to setup a single FXS line I have been
following the
instructions for Telephony Card Drivers on the
asteriskdocs.org site.
I have managed to
Hello,
I would like to ask, if there's a way to transfer a call from some
external program? I would like to build something like Asterisk Flash
Operator Panel, with the ability to transfer a call using drag and drop.
So I would like to connect to asterisk command line interface and
transfer one
Quoting Moises Silva [EMAIL PROTECTED]:
I downloaded and compiled this trunk version - Asterisk SVN-trunk-r28970. The
DTMF events show up in the logging system after I configured logger.conf to
output them, but they are not showing up in the Events.
On checking the SVN for the 6082 patch I saw a
Hello,
does anybody know about an existing skill-based routing solution for
asterisk? I found only some theoretical documents on voip-info.org.
I would like to have finer control over who can get which call in which
order.
Example:
Several operators with several topics.
Each operator may have a
2006/5/17, Cory Andrews [EMAIL PROTECTED]:
I think around Q3/Q4 of this year, you'll see some very interesting newproducts which incorporate DECT for wireless.For consumer products withlimited mobility, it seems to make a bit more sense than WIFI.
Cory,Which DECT products are you specifically
2006/5/16, [EMAIL PROTECTED] [EMAIL PROTECTED]:
Hi,I have got hold of a Nokia E60 myself but could yet connect to any SIPserver. WLAN works fine and i am able to browse thru my SMC wirelessLAN. If some one has successfuly done the SIP configs, i would like to
get the detailsThanks in
Simple solution. Use different queues for skillsets, add agents with
penalties based on their skillset. 1 means expert, 100 means you just
want the call answered if nobody with a lower penalty is available. Use
an IVR to direct the caller to the correct queue (topic).
Penalties can be set
Wow. That certainly looks like a definitive solution.
I'd have thought that any PC & soft phone with cheap USB audio I/F would provide at least simplex audio. I've seen USB I/Fs with XLR +4 dbm output at music shops for $40.
Michael
--Original Message Text---
From: Garth Summey
I need some help getting our DIDs updated in the
directory assistance and also the caller ID cname that is displayed. Does
anyone know where to go to do this so the major carriers will get this info? I
have found www.listyourself.net but I am
hesitating to submit info to someone I have
HmmmI have a 480i-CT. Does this mean that I might be able to add third party DECT handsets? Or just the matching Aastra handsets?
Michael
--Original Message Text---
From: Dovid Bender
Date: Sat, 20 May 2006 12:54:54 -0700 (PDT)
Cory,
Do you have the Nokia E70 and or the E60 ? If
I know it supports additional Aastra handsets, up
to 10 I believe. As for 3rd party, not sure about that...
Cory J AndrewsVOIPSupply.com454 Sonwil
DriveBuffalo, NY 14225++voice - 716.630.1555
X22email - [EMAIL PROTECTED]AIM - B2CORY
- Original Message -
On 19 May 2006, at 17:05, Mark Phillips wrote:
Hi folks,
With thanks to Alison Keenan (another Alison!) for the voice, Chris
Bagnal for converting from 44k wav to sln and finally Terje Elde for
debugging my HTML code, the British English files are now ready for
download.
They can be got from
I downloaded the SLIN and I have a couple of remarks.
1) you can't use the SLIN directly on a non-intel machine -
you may have to byte swap it first (took me a while to work
out why I just got pulse modulated static on my NSLU2 home
asterisk! (armv5teb) )
Can you explain in a bit more
If you need g729 and g723 format, let me know and i could convert it to you.
Tim Panton escribió:
On 19 May 2006, at 17:05, Mark Phillips wrote:
Hi folks,
With thanks to Alison Keenan (another Alison!) for the voice, Chris
Bagnal for converting from 44k wav to sln and finally Terje Elde for
On Sun, 2006-05-21 at 14:28 +0200, Olivier Krief wrote:
Hi,
I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you
help ?
From
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2,
I got the following:
1. Copy the desired
Does anyone have an idea how to limit the number of outging calls on a
sip trunk . limit=x only works for incoming calls.
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Juraj Bednar wrote:
I would like to ask, if there's a way to transfer a call from some
external program? I would like to build something like Asterisk Flash
Operator Panel, with the ability to transfer a call using drag and drop.
So I would like to connect to asterisk command line interface
On 21 May 2006, at 16:53, Chris Bagnall wrote:
I downloaded the SLIN and I have a couple of remarks.
1) you can't use the SLIN directly on a non-intel machine -
you may have to byte swap it first (took me a while to work
out why I just got pulse modulated static on my NSLU2 home
asterisk!
Does anyone have an idea how to limit the number of outging calls on a
sip trunk . limit=x only works for incoming calls.
__
Hi,
in the context where you dial out from:
exten = _X.,1,Set(GROUP()=OUTBOUND_GROUP)
exten =
At 08:37 AM 5/21/2006, you wrote:
I know it supports additional Aastra handsets, up to 10 I
believe. As for 3rd party, not sure about that...
Sounds better than it is, unless something's changed you can have 10
handsets but only 2 or 4 active calls.
Ira
Obviously a Radio 1 listener.
2) I was surprised to find that I didn't like the results.
This is a purely personal thing, but I found
Alison Keenan's delivery too redolent of a England that is
gone. I instantly felt like a child again, being told slowly and
clearly what to do.
2006/5/21, Greg Oliver [EMAIL PROTECTED]:
You have to upgrade to a new version of SCCP or older version of SIPbefore the bootloader on the phone will be able to handle the newerfirmware.In the same Cisco page you read the info is there - you can
either use an older version of SIP first, or a newer
Hi,2006/5/17, Rich Adamson [EMAIL PROTECTED]:
(*) By fax doesn't hangup, I mean though Asterisk server forward an incoming fax call to the right extension, it keeps on ringing the fax machine which never hangup. Maybe the flash signal is too weak
I'm very confused by the above statement.What do
2006/5/21, Steve Underwood [EMAIL PROTECTED]:
Lee Howard wrote: Olivier Krief wrote: For example, it seems that Brother 8360P uses Super G3 mode. Is there a fax-modem offering such capability so that I could easily check if I still cannothangup when I enable or disable Super G3 mode ?
MultiTech
Are any of these FCC licensed for use in the USA. DECT in the USA is
VERY new.
Michael Graves wrote:
HmmmI have a 480i-CT. Does this mean that I might be able to add
third party DECT handsets? Or just the matching Aastra handsets?
Michael
--Original Message Text---
*From:* Dovid Bender
Olivier Krief wrote:
2006/5/21, Steve Underwood [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Lee Howard wrote:
Olivier Krief wrote:
For example, it seems that Brother 8360P uses Super G3 mode.
Is there a fax-modem offering such capability so that I could
easily
Has anyone setup a n2p account into asterisk?
___
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mohamed kerbachi wrote:
Hi,
I have a TDM400P and it works,
Send us the output of your dmesg command.
Regards.
--- M.Hockings [EMAIL PROTECTED] a écrit :
In my attempt to setup a single FXS line I have been
following the
instructions for Telephony Card Drivers on the
asteriskdocs.org site.
I
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