Zapbarge only allows monitoring of a call.
As I understand intrude should allow 2 way audio.
I think maybe some trick with moving the call into a MeetMe conference room
could work.
Alfred.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of usedcanon
Sent:
I ran a PRI DEBUG SPAN 1 on our office system.
I could not see any FACILITIES messages on outgoing calls over the PRI.
So I suppose * does not send the CNAME messages at all on outgoing calls.
CLID NAME is just a subset of the generic user to user messaging on ISDN
networks.
It should be
I am very interested to try it.
Regards.
Alfred R. Nurnberger
F L O S Y S
Making Communications Flow
US Tel:+1 (360) 816-8800
or +1 (503) 972-9300
UK Tel: +44 (118) 321-6304
DE Tel: +49 (911) 3083-9316
FWD #: 271604
Fax:+1 (360) 816-8809
US Toll Free: 1-877-4FLOSYS
http
sipgate.de has DIDs in Germany and the UK.
-Alfred
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stephen
Karrington
Sent: Friday, April 09, 2004 4:08 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Who has access numbers in the UK and Germany?
I am looking for a ISDN BRI card (u-INTERFACE) to connect * to a US 5ESS
switch (Qwest).
According to Qwest they support CNAME delivery on their 5ESS switches.
Does * chan_capi support CNAME ?
Regards.
Alfred.
___
Asterisk-Users mailing list
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Title: RE: [Asterisk-Users] Need a list of asterisk built-in variables
How
about setting up a database table and use the CALLERIDNUM as search criteria for
the table
i.e. on the management console do:"
database put CLIEXT 5551212 121 "
in
your extensions.conf:
exten
=
Try
exten = s,1,Answer
exten = s,2,SetCallerID(0${CALLERIDNUM})
-Alfred
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matteo
Rancilio
Sent: Monday, March 22, 2004 4:12 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] setvar CALLERIDNUM
Matteo
ADSI is a slow inband protocol.
You will notice that when pressing a key in voicemail that the system
responds immediately but updating to a new screen takes a couple of seconds.
This delay is caused by the downloading of new screen data from *. ADSI is
based on the Bellcore caller id specs. So
The procedure you described is the reset procedure.
If the phone is locked then this will not work.
I have one of those. No chance to upload scrips on it.
According to Cheryl Millosi from Sayson there is no way to program it
without knowing the password.
-Alfred
-Original Message-
From:
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista
Sent: Friday, March 19, 2004 3:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ADSI slow?
Alfred R. Nurnberger wrote:
The procedure you described is the reset procedure.
If the phone is locked
Try this:
exten = _0.,1,Dial(Zap/g2/${EXTEN})
The dot at the end means that more digits may follow.
This way * determines end of number by a timeout.
In general you could use:
exten = _X.,1,Dial(Zap/g2/${EXTEN})
This way ANY length number would be sent out 1:1
Only drawback is the 5
Filtering the audio will not help much, especially not for the party on the
other end.
As already mentioned several times, the hum is caused by a imbalance of Tip
and Ring.
Try to a 100kOhm potentiometers with 10kOhm / 1W resistor in series and
connect it first between Tip and Ground, try to
The correct way to hide your callerid on a PRI interface is to set the
presentation indicator.
Some CO switches do a basic sanity check on the callerid they receive. If
you set the number string to empty
but the presentation indicator to allow the number they will replace the
number string by your
You
are right, Brazil uses DTMF caller ID.
The
format is very simple
Dtmf-DNUMBERDtmf-C
Asterisk has all the tools available to get DTMF caller
ID to work. (DTMF decoder routines,etc.)and T1-CAS uses a very similar
format.
I
guess somebody just needs to spend the time and programm it
Steven.
I played a bit with the distinct ring function and noticed that
* doesn't detetect disctinct ring on the very first ring.
Check your log and you will see that the distinct ring output is 0,0,0
After the 2nd or so ring the actual distinct ring pattern shows up.
So what happens is that on
Mike
A fractional T1 PRI setup like yours would look like
8 * B-voice + 1 * D-PRI + 15 * data
A full T1 PRI has 23 B-channels + 1 D-channel
You loose one timeslot compared to a regular T1 but you get
23 fully transparent 64 kbit/s bearer channels instead of 24 * 56kbit/s
ones. No
interface card which allows to daisy chain
several GSM(or TDMA or CDMA) units together to the T1/E1 master unit.
P.S: Our FXS module uses the same chipset as the Digium TDM400P card.
Regards.
Alfred R. Nurnberger
_
F L O S Y S
Making Communications Flow
Tel: +1 (503) 972-9300
Fax: +1 (503) 972
polarity reversal (wink pulse after seizing the line) and
constant reversal after answer.
Make sure that Ring/Tip is not reversed. DID lines need proper polarity to
work correctly.
Regards.
Alfred R. Nurnberger
_
F L O S Y S
Making Communications Flow
Tel: +1 (503) 972-9300
Fax: +1 (503) 972
Roger.
Quick and simple answer.
Yes.
-Alfred.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Roger
Schreiter
Sent: Monday, January 26, 2004 1:54 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Wildcard X100P usable in Germany?
Hi,
can I use the
It's NI-2
Yes it does.
Alfred.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Welter
Sent: Friday, January 23, 2004 12:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] MI2
My CLEC just called and asked if we will support the MI2 protocol on
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 3:48 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Re: Digium X100P for $43
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
EM is the traditional protocol to connect two COs or PBXs
together.
EM trunks are inherently symmetrical. So there is no subscriber or
CO side.
You
have to see about the higher level signalling protocol to make sure dialed
digits etc work right.
Regards.
Alfred R. Nurnberger
F L O S Y
Yes it does.
Regards.
Alfred R. Nurnberger
F L O S Y S
Making Communications Flow
Tel: +1 (503) 972-9300
Fax: +1 (503) 972-9309
US Toll Free: 1-877-4FLOSYS
http://www.flosys.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Welter
Sent: Tuesday
Steve.
You are saying this from your view of 2004.
But at the time R2 was developed there were no microcontrollers and tones
were decoded with LC filters.
R2 provides interactive capabilities base on a simple tones protocol to
retrieve ANI, dialed numbers,
signalling status etc. It's compelled
: Re: [Asterisk-Users] Zone Paging
Alfred R. Nurnberger wrote:
There are a number of paging interfaces available which connect to a
regular
phone line on one side
and to a paging amplifier on the other side.
Could you provide a pointer?
The search terms pager and telephone together
.
Regards
Alfred R. Nurnberger
---
F L O S Y S
Making Communications Flow
Tel: +1 (503) 972-9300
Fax: +1 (503) 972-9309
US Toll Free: 1-877-4FLOSYS
h323: 208.187.136.227
http://www.flosys.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Here are just a few examles for clarification.
Lets assume my local area code is (212)
--- local calls - usually free --
555-= local call 7-digit dialing area
212 555-= local call 10-digit dialing area
toll calls - metered
TDD is a very simple teletype like unit for Telecommunications for the
Deaf
Which is hooked up to a telephone line with an acousic coupler remember
these ?
It transmits with 45 baud / BAUDOT code , but unlike regular modems the
carrier is removed once the key has been released.
TDD is supported by
My guess would be that the NAT firewall times out and closes the port.
Reopening it from the inside is no problem, but access from the outside gets
blocked.
In order to keep the path open both ways, the client needs to send some kind
of messages with the proper IP/port in regular intervals.
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