Displaying 20 results from an estimated 3000 matches similar to: "Billing Telephone Number (BTN)"
2008 Feb 21
0
HoldMusic Beep
Does anyone have a audio file they would be willing to share for on hold
music?
I am looking for something like the old norstar beep every few seconds.
I tried 3 seconds silence, beep.wav, beep.wav. But it just didn't sound
right. I need one that has a "softer" beep.
Thanks!
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Forrest Beck
IAXTEL: 17002871718
jonforrest.beck at gmail.com
http://www.shift8.biz
2003 Oct 14
1
Newbie with questions
I have an existing Meridian PBX system that I am looking at replacing with Asterisk for the multi-office features. One of the many areas that I am unclear on is if we can use the existing phones from the old PBX system? Some are Meridian M7310 and some are Norstar M7208 models.
They each support multiple lines with a multi-line display, but I haven't seen anything on this list about sporting
2006 Jan 03
2
integration with Meridian/Norstar ATA2
I'm attempting to use an asterisk box with a Digium TDM01B as voicemail
for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're
having problems where hangup is not always (but sometimes) detected.
It's not detected probably 70% of the time or more. (The users transfer
callers to an extension--caller then has to navigate a menu to get to
the appropriate user).
After
2007 May 03
3
SIP RealTime Friends
I setup sip realtime. Is it possible to use a type of friend? User
and Peer seem to work fine.
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Forrest Beck
IAXTEL: 17002871718
jonforrest.beck@gmail.com
2007 Feb 08
3
Automatic Dial, Play message
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
For example, you have a database of
FirstName LastName PhoneNumber
Jon
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Forrest Beck
IAXTEL: 17002871718
jonforrest.beck@gmail.com
2003 Jul 23
2
Integrate Asterisk with Meridian phone system
Has anyone integrated Asterisk for Voicemail with a Nortel Meridian
phone system or know if this could be done?
Thanks,
John Haigh
2007 Feb 25
1
Marks SNMP HowTo
I followed Marks SNMP howto on Voip Magazine and ran into a small
problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/)
When asterisk is running as a non-root user (asterisk) SNMP request
for for the Asterisk MIB tree return nothing. If I quit asterisk and
run it as root, all is fine. Does anyone have a idea what is going
on? I have never used agentX, so I am unsure of what it is
2007 Mar 30
1
Paging
First off, A lot of thanks to this list. I have learned ton from
reading through the posts this past year.
I need some advise.
I have two group of phones connected to a single server.
Group1= SIP/2503&SIP/2504
Group2=SIP/3501&SIP/3502
I'd like to be able to dial an extension and page a certain group of
phones only if ChanIsAvail returns 1.
I am not sure how to go about
2005 Aug 10
1
PRI dropped calls w/ asterisk dropped between pstn & norstar
Hi,
I dropped an asterisk server with a TE405P between a Norstar Meridian
PBX and it's PRI PSTN connection. Everything seemed to work fine
using a pass-thru-type dialplan configuration... except now we've
realised that outbound calls to celphones get dropped upon connect,
but not on every call (almost always the first try, but not the
second). All other calls to non-celphones had no
2007 Apr 04
1
Polycom
I know this doesn't belong on this list but... I am looking to see if
anyone is using Polycom and knows of a web based software for
creating/managing the cfg files for polycom phones. I see that the
AsteriskNow will add provisioning support for Polycom phones. Since
it is still in beta, I was just looking to see if there was anything
else out there.
Thanks!
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Forrest Beck
IAXTEL:
2006 Nov 13
0
Native TDM Bridge
I have a two port TE205P Digium card. I have set everything up to
create a native zap bridge between the two spans. Everything works
perfectly except one thing. Our telco has a "password" that has to be
entered as soon as a long distance call is made. So if I dial a long
distance call from my meridian system, asterisk bridges the call
between two channels, my telco picks up and gives
2007 Mar 26
1
Server Recomendation
I am looking to install a system with 200 phones (polycom). There
will be about 30-40 simultaneous calls. I am looking at the Dell 1950
with Quad 2.66, 2Gig RAM, Two 160 Gig SATA Drives (Mirrored with a
Perc5 card), Dual Gig NIC, and RHEL 4.0. I will use two "gateways"
for my PRI's and FXS Cards so PCI won't be used. I will probably use
a small 14" 2U server to handle
2007 Apr 19
3
Outgoing CallerID
I am not sure of the best way to do this, so I thought I would query the list.
I have about 100 internal extensions ranging from 2000 - 2100. Each
internal extension has a external DID number. For example: 2001 =
5552871620. As you can see the internal externsion and DID don't
match in any way. What would be the best way to set the DID for when
a extension dials out on the PRI? In
2007 May 03
2
zttranscode crashes server
I was just looking to see if anyone else has seen this problem as well.
When asterisk starts up it loads the zttranscode module. The problem
exist when I use the init scripts to stop asterisk and then use the
zaptel init script to unload modules. Since the zaptel init script
didn't load the zttranscode module it will error out when trying to
unload the modules.
I built
2004 Dec 29
9
IP Phone recommendations?
Hey gang,
I'm looking at escaping from a Nortel Meridian CISC system to
Asterisk/Digium/SIP phones. I'm currently in the testing and proof of
concept phase. I'm going to need a SIP phone and don't want to
re-purchase and have "orphans" around.
We currently run Nortel 7310 phones and they work great.
I'm sort of overwhelmed by all of the different IP phones. I
2007 Mar 01
2
Asterisk 1.4.1
Any idea when 1.4.1 will be available. There is a bug fix in the cvs
head that I need, and I don't want to run the cvs build on a
production machine.
Thanks...
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Forrest Beck
IAXTEL: 17002871718
jonforrest.beck@gmail.com
2004 Jul 22
2
Nortel SL1 protocol and *?
I have been investigating more tight integration between * and the Nortel
MICS... it appears that it is at least theoretically possible to have *
store voicemail and log which stations call where.
Both require a T1 card. The T1 card requires either a clocking module or the
6-port fiber module to provide T1 timing. Naturally a T100P or TE405P is
required on the * side.
To log which
2007 Apr 24
2
Voicemail on Different Server
I have two seperate systems at two different locations. Each hosts
there own voicemail for their phones.
I have thought about just having all voicemail on one server. Is the
best way to do this just through a dial app?
For example, if someone dials 1000 to check voicemail at site A. The
dialplan will be something like this on Site A:
[context-for-phones-at-one-location]
exten =>
2007 May 05
1
ODBC
I am trying to compile asterisk with ODBC support on CentOS 4.4. I am
running into the same issue as documented in this bug.
http://bugs.digium.com/view.php?id=8214
The server is a Dell 2950 with Dual Core /64bit processors (2Gig RAM).
I tried creating a symbolic link link mentioned in the bug report,
but didn't have any luck.
Any one else had this issue? What did you do to get around
2007 Jul 15
1
TimeStamp a Recording
Has anyone come up with to timestamp a Recording? I am using a pretty
simple dialplan to record a audio file for a hotline. I'd like to
store the date and time it was recorded somewhere, Ast DB or MySQL DB.
Then when the audio file is played back to a caller, the system will
say something like.
This message was recorded
January
14th
at
10
42
pm
Thanks for any ideas you may have.
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