Displaying 20 results from an estimated 10000 matches similar to: "Analog DID"
2008 Feb 08
3
Question about Asterisk versions (newbie)
Hello,
I would like to consulate with you guys. I'm setting up an Asterisk
server on Debian. The problem is that Rhino drivers are only compatible
with Zaptel 1.2. By default debian stable offers asterisk 1.2 and zaptel
1.2, and that suits our needs. Is there a bleeding need to use latest
version of asterisk? I have managed to install Asterisk 1.4 and Zaptel
1.2 but then i got the
2008 Mar 06
14
FXS channel banks
Greetings list,
I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at.
If anyone's had experience using channel
2008 Jun 13
1
PRI crashing Asterisk
I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and
1.4.20 as well as the latest libpri no change
Progress is as follows......
< Supervisory frame:
< SAPI: 00 C/R: 0 EA: 0
< TEI: 000 EA: 1
< Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
< N(R): 025 P/F: 1
< 0 bytes of data
-- ACKing all packets from 24 to (but not including) 25
-- Since
2008 Feb 27
3
Simultaneous Inbound and Outbound calls on analog lines...
Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this
2008 Feb 20
2
Skype Users
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
found this today, I am not a skype user but have read on chan_skype
and don't like aspects of how it is implemented. My thoughts on it are
only theoretical as I haven't used it I just cringe at adding X to a
server. Anyhow there is a new project called sippyskype that appears
to do a similar sort of thing with a couple differences.
1. Its
2010 Jan 14
2
Dahdi issues
Hello,
My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port
modular card and a single FXS module.
Got the Rhino card installed and the machine sees it:
root at pbx:/etc/dahdi# dmesg | grep rcbfx
[ 71.985309] rcbfx 0000:04:00.0: PCI INT A -> GSI 21 (level, low) -> IRQ
21
[ 71.985440] rcbfx 1: Rhino PCI BAR0 50100000 IOMem mapped at
ffffc90008d7c000
[ 71.985504]
2009 Oct 02
1
Creating a clear channel on zaptel
Hi,
Is it possible to create a clear zaptel channel which doesn't require to be
picked up? The requirement of my client is to open a clear channel to a
recorder which starts recording certain message. Currently the channel which
is created by zaptel requires the other end to answer the call, and the
recorded can't answer, so the channel get hung up after a certain number of
rings.
Zaptel
2008 Mar 31
7
Cisco 7965 SIP Firmware
I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S).
Does anyone have a valid XMLDefault.cnf.xml they could share?
I have tried the version at
voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for
the 7941/7961 but unfortunately /var/log/messages shows
in.tftp stops sending after
2016 Apr 29
1
my dahdi dont'n start
<!DOCTYPE html>
<html><head>
<meta charset="UTF-8">
</head><body><p>Hello,</p><p>I have not resolved my problem.I renamed my dahdi file "mv dahdi.bash dahdi " in the directory /etc/init.d, but it doesn'nt work yet.</p><p>the same error after the command <strong>/etc/init.d/dahdi
2016 Apr 26
3
my dahdi dont'n start
On Tue, Apr 26, 2016 at 11:07 AM, Administrator TOOTAI <admin at tootai.net>
wrote:
> Le 26/04/2016 17:23, Mamadou NGOM a ?crit :
>
>> Hello,
>>
>>
>> Having installed DAHDI to be able to use the meetme() application , when
>> I start the dahdi service it generates me the following error:
>>
>> -bash: /etc/init.d/dahdi: No such file or
2008 Feb 18
1
Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)
Hi all... I did some Google searches and didn't find any info on this so
I'm posting it here... if this was recently discussed, I apologize for the
duplication -- please point me to the appropriate thread.
System Description:
Supermicro SuperServer 5015M-MF w/ PDSMi Motherboard
Intel Pentium 4 2.8 Ghz CPU
2 GB DDR2 Memory
Digium T400P 4 Port T1 Card
CentOS 5.1 (Final)
Kernel:
2007 Sep 12
1
Direct dialing to correct extension from analog lines
Hi,
I have a problem with people that are calling from analog lines.
We have a block of numbers 12345 - 0 to -99. Most calls are transmitting
the whole number including the extension. There's no problem with that.
But people calling from analog lines are connected to our asterisk box
as soon as they finish dialing 12345. They don't get a chance to dial an
extension.
Just inserting a
2007 Nov 26
2
Possible Conflicts with Junghanns 4 Port BRI and 8 Port Sangoma Analog in Same Box?
I know it is a strange arrangement but due to contracts, it is what it
is, no PRI for now.
I wonder if anyone on the list has run a server with both types of cards
installed? Results?
I have never touched a BRI except in concept and Cisco lab. Not sure
what the BRI stuffed package may or may not do to anything else that
might relate to zaptel or Sangoma.
Thanks,
Steve Totaro
2007 Jan 26
1
Analog FXO status checking
Hi all,
I would like to make a script/program that would be able to show lots of
status information from my analog FXO lines (and FXS lines in the near
future).
Example of interesting status information:
- Hook status: is there a call being made with that zap?
- Voltage status: cable connected, voltage values (if possible), line
ringing?
- RX/TX Volume status
I'm using a TDM400 card with
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
2007 Feb 28
1
Timing, use analog card, ZT Dummy etc.
Hello, we are setting up another system that will run either 1.2.4,
the latest version of 1.2 or 1.4. We have not yet decided on the
version.
Anyhow, this is a higher volume system (dual processor) which will
handle 30-50 simultaneous calls with 60 to 100 simultaneous channels
lit up. Most calls are g711 with very little g729 and a little gsm
mixed in.
We have a similar system doing exactly
2009 May 26
5
Maximum cable length for analog phone from FXS port
Hello.
I am looking for details of the maximum allowed/usable/effective wire/cable
length of the connection from a FXS port of Digium analog cards to the
analog telephone handset.
To clarify my intention, I need to have an analog telephone connection to my
asterisk box that is 3000 meters (3km) away at least. If you have any
details of ATA boxes or other similar devices that I could use to
2009 Jun 22
4
Different inbound routes for each interface on a TDM800P card.
I'm new to Asterisk and inherited this project so I apologize if this
question has been asked a hundred time before. I did start with Google
but I may not be asking the right questions, because I wasn't finding
any answers.
I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium
TDM800P to interface with our six analog phone lines from the telco.
Currently I have a single trunk
2007 Mar 24
2
TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC
Hi, everyone:
I am developing a system using Asterisk, TDM-400 analog cards, analog
lines, and Polycom SIP phones for internal extensions.
Initially there was bad echo but after a series of efforts, I've managed
to reduce it to a negligible level (it only happens when both parties
speak simultaneously, and even there, only for a few hundred
milliseconds). From an echo standpoint, things are
2009 Oct 08
4
No sound on voicemail from analog line
Hello.
I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound.
What can cause that problem?
Thanks in