On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
2009/12/4 Olivier oza-4...@myamail.com
Trying with a Junghanns PCI OctoBRI, I've got :
# dahdi_hardware
pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card
My initial thought was that wcb4xxp driver could not
Hello
Unless I overlooked it, the Asterisk Reference Information
Version 1.6.1.6 at www.asterisk.org/docs doesn't include instruction
on how to start Dahdi when used to drive a TDP PCI card (OpenVox A400P
with a single FXO module
www.openvox.cn/products/show.php?itemid=20lang=2).
I'd like to
It looks like make config takes care of installing an init script,
so I can just run /etc/init.d/dahdi start to load the required
modules.
I get the following error, however:
---
# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
wcfxo: [ OK ]
Running dahdi_cfg:
... but ls -l /dev/dahdi/ doesn't return channel #1 :-/
# ls -l /dev/dahdi/
total 0
crw-rw 1 root root 196, 254 Dec 8 13:38 channel
crw-rw 1 root root 196, 0 Dec 8 13:38 ctl
crw-rw 1 root root 196, 255 Dec 8 13:38 pseudo
crw-rw 1 root root 196, 253 Dec 8
Of course, as long as your endpoints support it. Read more about it
and purchase G.729 channel licenses for Asterisk from Digium:
http://www.digium.com/en/products/g729codec.php
Once you have the codec properly installed, enable it for your peer in
your iax.conf file allow=g729. Restart
I got it figured out: Modules must be listed in /etc/dahdi/modules:
wcfxo
wctdm
dahdi
/etc/init.d/dahdi start
dahdi_cfg -vvv
HTH,
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
2009/12/4 Olivier oza-4...@myamail.com
Trying with a Junghanns PCI OctoBRI, I've got :
# dahdi_hardware
pci::08:00.0 qozap- 1397:16b8 Generic OctoBRI ISDN card
My
Hi List!
I am running 'Asterisk 1.4.22 built by root'
I have an issue with voicemails. In the
var/spool/asterisk/voicemail/default/ext/inbox the msgnnn.txt files are
flagged as readable, this causes asterisk to just skip over the voicemails when
listening.
drwx-w 2 asterisk 4096
Hi List,
Apologies if this appears twice.. Apple mail seemed to post a follow up last
time that isn't appearing..
I am running 'Asterisk 1.4.22 built by root'
I have an issue with voicemails. In the
var/spool/asterisk/voicemail/default/ext/inbox the msgnnn.txt files are
flagged as
Hi -
I am having echo issues on our Asterisk box using a PRI circuit. I was
using the software echo cancellation and that helped a bit but didn't solve
it completely. So I went and bought a Digium echo cancellation module for
the TE121 card. That made it even worst, getting more echo on
On Tue, Dec 08, 2009 at 03:47:52PM +0100, Olivier wrote:
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
2009/12/4 Olivier oza-4...@myamail.com
Trying with a Junghanns PCI OctoBRI, I've got :
# dahdi_hardware
Hi List,
Apologies if this appears more than once.. Apple mail seemed to post a
follow up last time that isn't appearing so I've moved to webmail to send..
I am running 'Asterisk 1.4.22 built by root'
I have an issue with voicemails. In the
var/spool/asterisk/voicemail/default/ext/inbox the
On Tue, Dec 08, 2009 at 02:15:40PM +0100, Vincent wrote:
Hello
Unless I overlooked it, the Asterisk Reference Information
Version 1.6.1.6 at www.asterisk.org/docs doesn't include instruction
on how to start Dahdi when used to drive a TDP PCI card (OpenVox A400P
with a single FXO module
On Tue, Dec 08, 2009 at 03:37:26PM +0100, Vincent wrote:
I got it figured out: Modules must be listed in /etc/dahdi/modules:
wcfxo
wctdm
dahdi
You actually only need 'wctdm' .
And in fact, you could have generated that file with:
dahdi_genconf modules
/etc/init.d/dahdi start
I call into a box running asterisk 1.4.27.1 - this works.
on that box I run the CLI and enter the command core show channels concise
initially I see the ALSA/default.. and all that which is correct.
I continue to speak and continue to do the core show channels concise.
I continue to see the
We got the last two.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
If you're an asterisk 1.6 user, and use the 'Directory application', have
you noticed that the first keypress is always missed if you press it during
the part of the announement where alison says using you touch tone keypad
If this includes you, have a look at mantis bug
Hello,
I can't get the sound over alsa to work with Asterisk.
My current version is 1.4.21.2~dfsg-3 running on debian stable.
All settings are the default ones with exception of:
/etc/asterisk/modules.conf:
load = chan_alsa.so
noload = chan_oss.so
/etc/asterisk/alsa.conf:
On Tue, Dec 08, 2009 at 06:25:46PM +0100, vitaminx wrote:
Hello,
I can't get the sound over alsa to work with Asterisk.
My current version is 1.4.21.2~dfsg-3 running on debian stable.
All settings are the default ones with exception of:
/etc/asterisk/modules.conf:
load =
All
This is a small issue that I stumbled onto that has to do with the
channel numbering on an E1 connection into an Asterisk Zaptel/DAHDI
system.
As most of us already know an E1 has 32 channels of which 30(1-15
17-31) are B-channels and 1 (16) is a D-Channel. The 32nd channel is
not presented
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Dec 08, 2009 at 03:47:52PM +0100, Olivier wrote:
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
2009/12/4 Olivier oza-4...@myamail.com
Trying with a Junghanns PCI
Andrew Latham wrote:
and an example of my first thoughts:
bchan=1-15
dchan=16
bchan=17-31
uchan=32
Well, you've missed an important point: the DAHDI drivers for E1 cards
would have to be modified to make this 32nd channel in each span
actually exist, before any configuration in
On Tue, Dec 8, 2009 at 2:58 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Andrew Latham wrote:
and an example of my first thoughts:
bchan=1-15
dchan=16
bchan=17-31
uchan=32
Well, you've missed an important point: the DAHDI drivers for E1 cards
would have to be modified to make this
[Dec 8 18:24:48] ERROR[10571]: chan_alsa.c:693 alsa_read: Read error:
Resource temporarily unavailable
I agree, this looks like some form of conflict for the sound device.
The first thing I'd suggest doing, is trying to reproduce the
error with a command-line tool, with asterisk out of the
Hi friends, I am about to install an asterisk server using a Sangoma A101DE
over a Dell PE 2850 Server but I have doubts about PCI requirements.
First I see at sangoma page that A101DE is PCI-Express (I think x1 for the
size of the connector)
And the specs for the PE 2850 is
For PCI-X
Andrew Latham wrote:
This is where my query lives... What if... Imagine 2+ E1s sharing
the first E1's D-channel for timing and some manufacturer thought
about selling some hardware that would allow the use of 32 channels on
the next E1 and so on. So something like dchan=16
On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote:
After pressing *1 console is not showing anything indicating that the call
is being recorded:
I find that I often have to adjust the featuredigittimeout setting in
features.conf, as users tend to take their time between the * and 1 keys
when
2009/12/8 Ricardo Melendez rmelen...@utep.com.mx:
First I see at sangoma page that A101DE is PCI-Express (I think x1 for the
size of the connector)
Yes, it is PCIe x1. There is an A101D wich is PCI(-X).
for PCI Express
one x4 lane width
one x8 lane width
I can connect the card to any
On Tue, 2009-12-08 at 14:47 -0300, Andrew Latham wrote:
As most of us already know an E1 has 32 channels of which 30(1-15
17-31) are B-channels and 1 (16) is a D-Channel. The 32nd channel is
not presented in Asterisk Zaptel/DAHDI. There are other
configurations but this is the most common.
Slightly OT?
A client has two offices in the Virgin Islands that MUST maintain data
connectivity, and there are no available leased line options to run
a P2P link between them.
To date, broadband Internet connections at both offices have been used
as the link, with a VPN tunnel, and phones in
On Tue, Dec 8, 2009 at 4:01 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Andrew Latham wrote:
This is where my query lives... What if... Imagine 2+ E1s sharing
the first E1's D-channel for timing and some manufacturer thought
about selling some hardware that would allow the use of 32
On Tue, Dec 08, 2009 at 06:51:12PM +0100, Olivier wrote:
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Dec 08, 2009 at 03:47:52PM +0100, Olivier wrote:
2009/12/8 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
2009/12/4
snip
A client has two offices in the Virgin Islands that MUST maintain data
connectivity, and there are no available leased line options to run
a P2P link between them.
snip
Is there line of sight? I've been wanting to do a long-shot wifi link and my
company would give it a shot if you want :).
2009/12/8 Joseph syscon...@gmail.com:
After pressing *1 console is not showing anything indicating that the call
is being recorded:
-- Executing [...@office-closed:1] Playback(SIP/479-1270-680060b0,
transfer) in new stack
-- SIP/479-1270-680060b0 Playing 'transfer' (language 'en')
Hi David,
On Tue, 8 Dec 2009, David Gibbons wrote:
snip
A client has two offices in the Virgin Islands that MUST maintain data
connectivity, and there are no available leased line options to run
a P2P link between them.
snip
Is there line of sight? I've been wanting to do a long-shot wifi
Actually yhe best one who answered me before is xavimes, but did not understand
well his explaination, so I am still searching and need a help.
The realm is like a domain and it is used for authentication, this kind of
authentication is used when we are going to register from a wireless phone
The echo between our extensions (using Polycom 550 handsets) disappears once I
removed the Digium echo module. We are still experiencing some echo on land
line calls, using dahdi to connect to our PRI circuit.
What kind of settings do you recommend for the txgain and rxgain? Do I make
the
On Tue, Dec 8, 2009 at 7:37 AM, Noah Miller noahisaacmil...@gmail.com wrote:
Hi -
I am having echo issues on our Asterisk box using a PRI circuit. I was
using the software echo cancellation and that helped a bit but didn't solve
it completely. So I went and bought a Digium echo cancellation
I can't seem to locate any documentation on what this does. I tested it
out with a simple static conference room:
exten = conference,1,MeetMe(,1aMqw)
and a static room defined in meetme.conf:
conf = 123456,22,1
Users can get in with either of the pins, but I don't see that it does
anything -
What you say...Hose (hose+aster...@bluemaggottowel.com):
I can't seem to locate any documentation on what this does. I tested it
out with a simple static conference room:
exten = conference,1,MeetMe(,1aMqw)
and a static room defined in meetme.conf:
conf = 123456,22,1
Users can get
Hi,
I have just recently been using DAHDI, and I wanted to know how to monitor
capacity.
Let's say I have two DS1 (23 channels) coming in, one for Florida (let's
say) and one for New York. How can I get a reading of how many channels of
each T1 port is being used at any given moment?
Core show channels shows all calls. you will get two entries for most
calls, 1 for the dahdi channel and one for the sip phone using it.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, December 08, 2009
From the CLI:
asterisk -rx 'core show channels' | grep DAHDI | sort -n
Channels with a value of 1-23 are on your primary DS1, channels with a value of
25-47 are on your second DS1.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Mike l...@virtutel.ca wrote:
Thanks Jared,
That solution was perfect!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith
Sent: 07 December 2009 14:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Thanks Tim and Danny. It seems a more direct way should be there, but that`ll
work.
Regards,
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Tuesday, December 08, 2009 16:45
To: Asterisk Users Mailing
I`ve just experience a dead server, because I ran /etc/init.d/dahdi restart
. I had to reboot the server.
Should I worry about something not being right in my install, or is there a
known problem with doing this while Asterisk is running?
I expected DAHDI channels to die, but not the
On Tue, 2009-12-08 at 19:04 -0500, Mike wrote:
Thanks Tim and Danny. It seems a more direct way should be there, but
that`ll work.
A more direct way would be to use SNMP in Asterisk and keep statistics
with Cacti. That way you will have an historical view of usage by hour,
day,
At 10:38 AM on 06 Dec 2009, Thomas Perron wrote:
I am trying to use a simple tool in the Dial plan so that if the first
number does not connect the logic will go to the second and/or third.
Basically, I want the call to ring and connect to the first number
Then, if it is not answered I
That`s my plan exactly, but for that I need some value to poll, and I was
looking for the most efficient way to know that 12 out of 23 channels are being
used.
Seems that I need to massage the data more than I wanted, instead of using a
dahdi show port 3 command. That`s what I meant by it
you have to stop asterisk before restarting dahdi service
On Dec 8, 2009, at 7:06 PM, Mike wrote:
I`ve just experience a dead server, because I ran /etc/init.d/dahdi restart .
I had to reboot the server.
Should I worry about something not being right in my install, or is there a
known
Have a trunk 1.4 asterisk, running on centos on the lan at work.
A long story, but we had the entire work network on a public address
range (90.1.0.x), going to a firewall, then out to the net.
At home (192.168.1.x network) I have a router that connects to the
firewall via a vpn tunnel.
All was
On Tue, Dec 08, 2009 at 07:06:52PM -0500, Mike wrote:
I`ve just experience a dead server, because I ran /etc/init.d/dahdi restart
. I had to reboot the server.
What version of DAHDI (tools, linux)?
What DAHDI hardware (if any) do you have? What do you have on
/etc/dahdi/modules ?
Should
On 12/08/09 11:11, Jared Smith wrote:
On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote:
After pressing *1 console is not showing anything indicating that the call
is being recorded:
I find that I often have to adjust the featuredigittimeout setting in
features.conf, as users tend to take their
What's the output of:
lspci -v -nn -s 08:00.0
# lspci -v -nn -s 08:00.0
08:00.0 ISDN controller [0204]: Cologne Chip Designs GmbH ISDN network
Controller [HFC-8S] [1397:16b8] (rev 01)
Subsystem: Cologne Chip Designs GmbH Device [1397:b552]
Flags: medium devsel, IRQ 10
I want to rebuild my mixmonitor file.But this time I just want the
recording is from the time when the client answer the call,not from the
beginning. Anybody can help?
Daniel
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Joseph wrote:
On 12/08/09 11:11, Jared Smith wrote:
On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote:
After pressing *1 console is not showing anything indicating that the
call is being recorded:
I find that I often have to adjust the featuredigittimeout setting in
There is another setting which I can't find at the moment which controls
this -- its normally set to 500ms.
Francesco Peeters france...@fampeeters.com wrote:
Joseph wrote:
On 12/08/09 11:11, Jared Smith wrote:
On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote:
After pressing *1
Hi,
In /var/log/asterisk/full (Asterisk 1.6.2-rc6), I can see :
[Dec 8 15:02:17] VERBOSE[10283] config.c: == Parsing
'/var/spool/asterisk/voicemail/default/103/INBOX/msg.txt': [Dec 8
15:02:17] VERBOSE[10283] config.c: == Found
[Dec 8 15:02:17] VERBOSE[10283] file.c: --
58 matches
Mail list logo
