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: -- Playing
'vm
There is another setting which I can't find at the moment which controls
this -- its normally set to 500ms.
Francesco Peeters 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" console
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"
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
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a
>
> 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
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
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 ?
>
> Sh
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 wa
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 problem
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 be
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
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,
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 whole
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 L
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: [aste
>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" wrote:
>
>
Hi,
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
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? Idea
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
>
> Use
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 -
On Tue, Dec 8, 2009 at 7:37 AM, Noah Miller 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 module for
>> the
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
th
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
(
Hi David,
On Tue, 8 Dec 2009, David Gibbons wrote:
>
> 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.
>
> Is there line of sight? I've been wanting to do a long-shot wifi
2009/12/8 Joseph :
> 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
> -- Playing 'transfer' (language 'en')
> -- Executing [...@office-cl
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.
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 :).
Do you
On Tue, Dec 08, 2009 at 06:51:12PM +0100, Olivier wrote:
> 2009/12/8 Tzafrir Cohen
>
> > On Tue, Dec 08, 2009 at 03:47:52PM +0100, Olivier wrote:
> > > 2009/12/8 Tzafrir Cohen
> > >
> > > > On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
> > > > > 2009/12/4 Olivier
> > > >
> > > > > Tr
On Tue, Dec 8, 2009 at 4:01 PM, Kevin P. Fleming 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 channels on
>>
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, 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
2009/12/8 Ricardo Melendez :
> 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 of the slots?,
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
whe
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
> bchan=1-15
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
> [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 loo
On Tue, Dec 8, 2009 at 2:58 PM, Kevin P. Fleming 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 32nd
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 chan_
2009/12/8 Tzafrir Cohen
> On Tue, Dec 08, 2009 at 03:47:52PM +0100, Olivier wrote:
> > 2009/12/8 Tzafrir Cohen
> >
> > > On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
> > > > 2009/12/4 Olivier
> > >
> > > > Trying with a Junghanns PCI OctoBRI, I've got :
> > > > # dahdi_hardware
> >
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
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:
>
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:
input_device=d
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
https://issues.asterisk
We got the last two.
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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
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
>
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 mod
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 03:47:52PM +0100, Olivier wrote:
> 2009/12/8 Tzafrir Cohen
>
> > On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
> > > 2009/12/4 Olivier
> >
> > > Trying with a Junghanns PCI OctoBRI, I've got :
> > > # dahdi_hardware
> > > pci::08:00.0 qozap- 1397:
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
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 readabl
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 2009
2009/12/8 Tzafrir Cohen
> On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
> > 2009/12/4 Olivier
>
> > 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 wcb4xx
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,
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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 a
... 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 13
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: DAHDI_CHAN
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=20&lang=2).
I'd like to
On Tue, Dec 08, 2009 at 08:50:40AM +0100, Olivier wrote:
> 2009/12/4 Olivier
> 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 support PCIe cards, a
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