Displaying 9 results from an estimated 9 matches for "jerimiah".
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jeremiah
2003 Nov 25
8
Prompt recording
Does anybody have useful tips on creating good quality recordings for
use with prompts in asterisk? I'm interested in hearing input on
hardware (mics, dats, sound cards, etc) and software (recording
software, dsp) as well as recording techniques.
Jerimiah
Tularosa Communications
2004 May 07
0
- Re: Routing by called interface - Email found in subject
...We
then also tie the calleridname to which channel they dial out from as
well.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jerimiah
Cole
Sent: Friday, May 07, 2004 1:01 PM
To: asterisk-users@lists.digium.com
Subject: [SPAM] - Re: [Asterisk-Users] Routing by called interface -
Email found in subject
Chris Wilson wrote:
> I want to run different lines directly to different extensions on two
> FXO analog interfaces. ie; Za...
2003 Oct 27
0
Asterisk on SPARC
Has anybody tried running Asterisk on a SPARC based system? I'd imagine
drivers would be the major issue. Any info is appreciated.
Jerimiah
Tularosa Communications
2004 May 07
3
Routing by called interface
Hey everyone,
I want to run different lines directly to different extensions on two
FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
extensions 102
Does anyone know of a way to do this?
Thanks!
Chris
2004 Jul 06
2
GR303
iH
where can i find documentation on Asterisk's support for GR303???
thanks
- hcir
2003 Oct 20
4
MOH different question
Is there anyway for a sip station to play MoH out of the speaker?
I know I can do it by calling the station and putting it on hold.
For example:
On a samsung pbx with MoH, if you goto one of the workstaions and press
a button
The moh plays out of the speaker.
Is there any way to do this with asterisk?
Kevin,
Honeycomb Internet Services
2004 Aug 20
4
Help with upgrading 7960 SCCP to SIP
I've got a Cisco 7960 that I'm trying to convert to SIP. Here's what
"Firmware Versions" says:
App Load ID:
P00305000300
Boot Load ID:
PC03M030
Version:
5.0(3.0)
The files that I have are:
P003-07-1-00.sbn
P003-07-1-00.bin
P0S3-07-1-00.loads
P0S3-07-1-00.sb2
If I put "P003-07-1-00" in OS79XX.TXT, the phone tries to tftp
XMLDefault.cnf.xml. I've
2004 Sep 29
5
music on transfer
Good day all
I got my Music on hold to work but can I/how do i get music on transfer?
Please help
Thanks
2003 Oct 24
8
SS7 signaling/Softswitch
I'm confused a bit about the following and was hoping to get some answers on
this group - What is exactly implied when we say asterisk can connect to a PSTN.
Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume
asterisk does not need to do any SS7 signaling and all it does (playing the role
of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct
statement?