search for: jerimiah

Displaying 9 results from an estimated 9 matches for "jerimiah".

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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?