Displaying 20 results from an estimated 900 matches similar to: "Transfer directly to voicemail (blind transfer)?"
2006 May 22
10
US telco lingo
Could someone explain to a non-US dummy the following phrases I have seen on
the list.
"I can provide you with tier 1 termination 6/6. I can blend or NPANXX
breakout."
"We provide US48 termination, blended rate for 1 MOU and above is .008 with
6/6."
What is 6/6?
What is US48?
What is blended?
What is MOU?
What is NPANXX breakout?
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2003 Aug 25
2
0 out of voicemail to different secretaries
Is it possible to configure * so that if a caller reaches voicemail for
someone in Engineering, but doesn't want to leave a message they can
press zero (0) and reach the Engineering Secretary or if they are
calling someone in Accounting and reach voicemail, pressing '0' would
reach the Accounting secretary, not the Engineering secretary?
Don Pobanz
2003 Nov 25
2
modem to modem calls through asterisk
Modem connect speeds on calls through * seem to be lower than calls
made through the telephone company lines or our old Rolm PBX. All data
calls have 2 wire analog modems on both ends.
For my set up I have channels of a Zhone channel bank tied to 2 modems.
The Zhone channel bank interfaces my * server with a T400P card.
modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -
2006 Nov 19
4
reduce dialtone volume on zap channel.
Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work.
I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized.
Any other thoughts on how to solve this are also
2007 Dec 26
2
No cdr_csv after upgrade from 1.2.x to 1.4.x
After upgrading from 1.2.x to 1.4.x call detail records are not being
written to /var/log/asterisk/cdr-csv/Master.csv
In cdr_manager.conf I have
[general]
Enabled = yes
Apparently there is something else that needs to be configured for call
detail records in 1.4.x. Can someone point me in the right direction?
Don Pobanz
2005 Jun 22
1
Dialplan Q: Dialing with Capi
Hello,
I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI
as channels. A call comes in via IAX2 and should be redirected to CAPI.
So I wrote the following dialplan:
[fromiax]
exten => _8XXX,1,Answer
exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r)
[fromcapi]
exten => 265,1,Answer
exten => 265,2,Dial(IAX2/PoC/11@from-lw)
exten => 265-BUSY,1,Busy
exten
2003 Jul 27
3
Nortel 350
Wondering, since they appear to be plentiful on eBay, whether I could
get a Nortel 350 to use to learn my way around ADSI.
The vendor claims that these are "generic," and looking through the
archives I wonder if that means that they might be unlocked in the sense
that the word is meaningful to asterisk.
Of course I am green as could be on this topic, so this question may
even be a
2006 Jun 08
2
no dialtone on channel banks
Hi all,
I am having a problem on to different boxes and to different channel
banks. I can't get dial tone out of either one. I can still send and
receive calls but no DT. This is the error that I get:
Jun 8 15:23:54 WARNING[5021]: chan_zap.c:6283 handle_init_event: Unable
to play dialtone on channel 95
asterisk-1.2.9.1
zaptel-1.2.6
zapata.conf:
signalling=fxo_ls
context=gene
2005 Mar 08
2
Please help with install * SOLVED
Thanks anyone, I found the problem in rhconfig.h.
After the fix I successfully compiled zaptel.
V.
--- Ron Wellsted <ron@wellsted.org.uk> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Have you built your kernel on that machine?
>
> The errors suggest that while the kernel sources are
> installed, the
> kernel has not been built.
>
> Check on
2003 Nov 05
4
error compiling asterisk
I did cvs update on asterisk, zaptel, libpri as of today (November 5,
2003). I also did 'make clean' on each of them. My previous version of
asterisk was cvs of September 15, 2003. No other changes have been made
to my system other that these updates.
when running
'make asterisk'
the following error appears
term.c:55: conflicting types for `term_color'
2007 Jan 05
4
how to transfer calls when analog phone has no transfer button
When you have a bunch of analog phones that you want to connect to
asterisk, but those analog phones have no transfer button, what are
the options to allow the phones to transfer a call?
--
------------------------------------------------------------
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507)
2006 Mar 28
4
ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
Did anybody know,
Is it possible to establish a ISDN DIAL up Connection and Analog Dial up
Connection (V90) trough asterisk with Digium TE405?
Thanks a lot for help.
Nico Giefing
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2004 Dec 28
2
Wildcard remote looping
Is there something special that needs to be done to allow a T100P/T400
to respond to a remote loop request?
I am having some issues with a Point to point T1 line using zaptel ppp.
This line gave us small problems when we had a pair of Cisco 2600's on
either end but now with the zaptel ppp it is going down every couple of
minutes for 15 intervals.
The phone company says the line is good to
2007 Oct 18
1
Limit number of times a call can be forwarded
We have had a few different times when a user has forwarded their phone
to himself. This has overloaded the communications to our operator panel
(FOP). One user should not be able to effect the whole phone system!
Is there a way that the number of times that a call can be forwarded
could be limited like to 10 or even 100? Then even if a user does
something stupid like forwarding their calls to
2003 Nov 04
1
asterisk and zplex10b (fwd)
hello all,
I still experience the random off-hook on my fxo cxhannels, i am using a
zplex-10b channel bank. which does not allow me to call out.
The situation still persists...this is what i have in the zapata.conf
[channels]
context=internal
context=incoming
context=default
usecallerid=no
usecallwaiting=no
signalling=fxs_ks
channel=1-8
signalling=fxo_ks
channel=16-24
but i still have the the
2004 Jun 14
4
Sipura 2000 not answering em_w calls
I recently purchased a Sipura 2000 and connected a phone to it which is
connected to my asterisk box via sip.
Calls to the Sipura 2000 work fine from another sip device connected
through *, from either an fxo or fxs (via adtran channel bank connected
to a T400P card) port. However, when a call comes in from the phone
company over a T1 with em_w trunks, the phone on the Sipura will ring
but I
2005 Oct 04
3
ADSI -- is it dead? Worth bothering with?
Since Colin Anderson -- in a previous thread -- asked the question about
whether ADSI was dead, I thought it was worth discussing.
We've been using Nortel Vista 350s in our office up until now. The
phones are from Telus, I don't know if there's any way to unlock them.
It would appear Telus hasn't done much in the way of updating software
for these phones; or if they have, they
2006 Feb 21
5
Voicemail 0 for operator call routing
Does anyone know of a way to specify what extension is dialed when 0 is
pressed in the voicemail system. I have a situation where there is more
than one secretary and they want the 0 to redirect to the appropriate
secretary for the two groups of people.
So an example would be:
555-1234 -> voicemail -> Secretary 1
555-1235 -> voicemail -> Secretary 2
Any help would be greatly
2004 Jan 11
6
T1 Sync clarification
Hi List,
After reading a bunch of the docs, list post archives, it
still seems that a clear definition of how to clock the T100P
card is muddy.
zttool says that the link is "INTERNALLY CLOCKED",
does this mean the T100P is providing clock, or does
this mean the T100P is getting clock from the T1 line
side (ergo getting clock from the Telco) ??
If you have sync = 0 then zttool
2003 Jun 30
1
T1 slips/BPVs clarifications (was: Help! Problems talking to upstream switch)
Just some clarification here.
On Monday, June 30, 2003 12:04 AM, Steven Critchfield
[SMTP:critch@basesys.com] wrote:
>
> As for slips and bipolar violations...
> T1s are just high speed serial lines. A sleep is when you loose sync
> with the far side and when you see a 1 come across the line, you may
> not
> know which bit it was for.
A slip is when one end of the circuit is