.
You may want to check your /etc/modprobe.d/dahdi.conf
Try loading the driver manually as follows:
1. Stop Asterisk
2. modprobe -r wctdm
3. modprobe wctdm
4. dahdi_cfg -vvv
Good luck,
Vladimir
On 9/23/2011 4:27 AM, Remco Barendse wrote:
Hi list
I have 2 servers with a TDM400 card, port 1
On Mon, 26 Sep 2011, Michael L. Young wrote:
I think the clue is actually right there in the error message.
You say that port 1 is an FXO module? Then your signaling is set wrong. The
signaling should be fxsks.
For port 4, it should be fxoks.
Remember, that in the configuration files,
Hi list
I have 2 servers with a TDM400 card, port 1 populated by an FXO (red)
module), port 4 populated with an FXS module. I am using dahdi
linux and tools 2.5.0.1. The servers are running CentOS 4 and the other
box CentOS 6.
Both modules have been working fine but recently stopped
I am reinstalling a server and wanted to give the asterisk rpm's a try.
It seems however that the repo on asterisk.org doesn't know anything more
recent than RHEL / CentOS 5.
Is there a more recent repo?
Will asterisk work with selinux?
Thanks!
--
Reading of all the brute force attacks on the list i was wondering if
anyone has implemented some connection rate limiting rules in Shorewall to
stop the brute force attacks?
I'm a bit puzzled about which rule(s) to use on which ports, if
anyone could help me with some example rules to start
On Sun, 11 Apr 2010, Mark Smith wrote:
Same this end from 184.73.17.150.
Use this little piece of iptables magic to block the whole of Amazon's EC2 ip-
range.
iptables -F
iptables -A INPUT -m iprange --src-range 216.182.224.0-216.182.239.255 -j DROP
iptables -A INPUT -m iprange
I currently have some Asterisk home servers on general pc hardware as well
as a mission critical server asterisk pbx running on a Dell 2850
To reduce noise and power consumption i would like to migrate them all to
an Intel Atom based solution, showstoppers so far were single NIC and
single PCI
On Mon, 18 Jan 2010, ev...@disruptor.nl wrote:
Hey Ira,
It seems after a several testing, that the wait(1) seems to solve the issue.
Only now weirdly enough the phone keeps ringing if the caller hangs up
before i picked up the phone (pstn call)
Regards,
Evert
Hi Evert
I have been
Is there any fix or workaround for the DNS problem (old standing bug that
when the box starts and domain names do not resolve quickly enough from
DNS then asterisk stops using the outgoing trunks.
I read on the list before that it is considered a huge and unacceptable
load for asterisk servers
I am using asterisk 1.6 at home and would like to send incoming calls
without caller id immediately to voicemail (i don't want to use the
privacy manager where people have to enter a number).
The config examples i found are all for the pretty obsolete 1.0 and 1.2
versions of asterisk.
Would
I never do sip show channels but i tried it this morning to see if
everything is working after upgrading 2 boxes to 1.6.0.19 and 1.4.27.1
Is it correct that Asterisk doesn't clean up sip channels anymore after
using them ?
On one box i can see sip lines for every phone that was attempted to
Thanks for this new release :)
Just out of curiosity, why did the download page for asterisk.org change?
In the old days it was quite clearly visibible what the latest asterisk
version was including all the related packages for that version
(libpri, addons, dahdi etc. etc.)
Why not do
On Mon, 2 Nov 2009, SIP wrote:
That's odd. We've had Snom 190s, 320s, and 360s running day in day out
for years with not a single issue. Maybe we got all the good ones from
your batch. If that's the case, I thank you for 'taking one for the
team' as it were. ;)
Perhaps i am REALLY unlucky
On Fri, 30 Oct 2009, hbk wrote:
Hi,
I have played with the 820 for some weeks, mostly love it excellent speech
quality. Even got the mini browser running
showing my favorite webcam, this could be put to real use too:)
Issues so far:
Some embarrassing crashes while speaking, was able to
I have a TDM400P hooked up to an analog line from KPN in The
Netherlands. CallerID is working but sometimes some digits are missing
from the number, i.e. if the number that calls me is:
0204569236
I will sometimes get this in the display:
020456236
Which digit is missing seems to be fairly
Hi list,
To make outgoing calls by skype i would like to have our crm app create
callfiles like we do for normal calls.
If i read the instructions it says this :
---quote---
The syntax for making an outgoing call using Skype for Asterisk is as
follows:
Dial(Skype/[originator@]destination)
On Wed, 2 Sep 2009, Matt Riddell wrote:
On 2/09/09 7:45 PM, Remco Barendse wrote:
So i create a callfile that looks like this:
---
Channel: SIP/228
MaxRetries: 0
Dial(Skype/asterisk...@somebodyonskype)
Priority: 1
Callerid: Somebodyonskypesomebodyonskype
You're combining technologies
On Wed, 19 Aug 2009, Terry Wilson wrote:
I haven't seen (or heard of) it happening. Please post a bug report
on http://betareports.digium.com/mantis/ with a backtrace from one of
the core dumps along with the relevant information about your setup,
dialplan, chan_skype.conf, etc. If there
On Fri, 21 Aug 2009, Olivier wrote:
So basically it's harmless, unless you actually have such a card.
Yes, but as you mentioned, most don't have a transcoder card.
My opinion is such message shouldn't be send at all for those environments
where there is no transcoder card, (as it will
I have a CentOS release 4.7 box running asterisk-1.4.26.1 with
dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0
I regularly get these messages, is this something i should be worried
about?
[Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so = (ITU
G.726-32kbps G726 Transcoder)
[Aug 21 01:05:07]
On Wed, 19 Aug 2009, Terry Wilson wrote:
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity to actually fix any bugs
that people find. :-)
I installed the 1.0 release of Skype for Asterisk and last
For Asterisk for Asterisk trunk.
Julian
2009/8/19 Remco Barendse aster...@barendse.to:
On Tue, 18 Aug 2009, Terry Wilson wrote:
That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.
Have you posted a bug describing the issues you are having
On Mon, 17 Aug 2009, Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
http://store.digium.com/productview.php?product_code=1SFA0001
Yes, pretty pricey indeed especially considering that you can buy Skype
ATA
On Tue, 18 Aug 2009, Terry Wilson wrote:
That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity
I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk
1.6.1.
As FreePBX only supports ZAP naming i set dahdichanname = no in my
asterisk.conf.
However, after installation the console was still merrily chattering about
incoming calls on DAHDI channels and nothing happened
On Thu, 18 Jun 2009, Kevin P. Fleming wrote:
Remco Barendse wrote:
I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk
1.6.1.
As FreePBX only supports ZAP naming i set dahdichanname = no in my
asterisk.conf.
However, after installation the console was still merrily
On Wed, 3 Jun 2009, Rob Hillis wrote:
Christian Stredicke wrote:
Check out the snom 300 or the snom 820...
Good lord... talk about two extremes... :) The Snom 300 is pretty good,
but the 320 is much better and costs around a *third* of what the Snom
820 does.
Stick with the older model
On Sat, 9 May 2009, Tim Panton wrote:
This is a bit off topic, because I 'think' it isn't an Asterisk problem.
However I'm not sure and anyhow I'm hoping someone may recognize the symptom.
We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years
old)
were packed up for
On Tue, 12 May 2009, Andrew Joakimsen wrote:
Overall, given the limitations of WiFi, it works rather well. I've
never had to reboot my E71 or play with the settings after it was
setup. Something I can't say about other WiFi (only) phones I have
used. And VoIP on Windows mobile phones is crap.
On Sun, 8 Mar 2009, benoit wrote:
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
The TE220P and the TDM800P are in different Asterisk boxes? Any particular
reason for that? I now have an E1 coming in to
Hi all
I don't know what went wrong but i no longer seem to be able to compile
asterisk. I first do :
cd /usr/src/dahdi-linux-2.1.0.4
make clean ; make all ; make install
cd /usr/src/dahdi-tools-2.1.0.2
./configure ; make clean ; make all ; make install ; make config
So far so good but
On Sun, 8 Mar 2009, Sebastian wrote:
The fax error seems to be problem of spandsp version.
What version are you using???
I use the latest IAXMODEM 1.2.0, the changelog of it says update spandsp
to 20080725 snapshot
However, i never asked Asterisk to compile with fax support, can i disable
On Sun, 8 Mar 2009, Tzafrir Cohen wrote:
On Sun, Mar 08, 2009 at 12:26:00AM +0100, Remco Barendse wrote:
So far so good but then when i do :
cd /usr/src/asterisk-1.6.0.6
make clean ; ./configure ; make all ; make install
i get this :
In file included from app_dahdiras.c:50:
/usr
On Fri, 6 Mar 2009, Klaus Darilion wrote:
Updating to 1.4 branch solved the issue. Thanks.
Pity that they still didn't release a new version that works properly.
1.6.0.6 is broken too, SIP doesn't work on 2 difference boxes i tried it
on.
___
--
I tried upgrading to 1.6.0.6 but when i compile and install that, it seems
that support for SIP is missing completely?
Reverting back to 1.6.0.5 gets SIP going again...
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On Tue, 10 Feb 2009, Steve Totaro wrote:
Kannel is probably the best way to go in the States, unless you want
to sign up with an aggregator.
I use Kannel and a bank of Sony Ericsson phones. To send SMS, you
just have to hit a URL on the Kannel server with a properly formatted
URL. I just
On Tue, 10 Feb 2009, Steve Totaro wrote:
In that case, you would not need Asterisk at all. If you can create
call files can you hit a URL from your CRM as well?
Not really, the app cannot open a browser, but it can create a file on a
samba share quite easily.
Therefore going through
1.4.23.1 is quite badly broken and there are no significant new
features
Better to revert back to 1.4.22.1
On Tue, 3 Feb 2009, Jose P. Espinal wrote:
Hello List,
I have been working on a little PHP software that uses AMI's
UpdateConfig command in order to modify some of it's config
1.4.23.1 doesn't seem to work for me.
I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest
zaptel as well. Incoming calls stopped working. Whenever an extension was
trying to pickup the phone by doing a group pickup with *8 the extension
just got dead audio and the next
On Thu, 29 Jan 2009, Thomas Stein wrote:
On Thursday 29 January 2009 09:23:41 Remco Barendse wrote:
1.4.23.1 doesn't seem to work for me.
I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest
zaptel as well. Incoming calls stopped working. Whenever an extension was
trying
Because of a kernel upgrade i needed to recompile DAHDI. Dahdi 2.0.0 with
2.0.1 was working ok, after the reboot and compile it doesn't start
anymore.
Firstly it tells me it is using /etc/zaptel.conf which is deprecated, if i
remove that file it complains that this file is missing.
On Sun, 18 Jan 2009, Tzafrir Cohen wrote:
Unloading DAHDI hardware modules: doneLoading DAHDI hardware modules:
wctdm: Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
grep ztcfg /etc/modprobe.conf /etc/modprobe.d/*
Remove those lines.
On Thu, 9 Oct 2008, Steve Totaro wrote:
I don't have answers just a question.
DAHDI is alpha or beta code, what motivates you to upgrade so badly that you
are frustrating yourself so much?
Perhaps the fact that zaptel is not listed anymore on the Digium website?
:)
On Thu, 9 Oct 2008, Sean Bright wrote:
On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse [EMAIL PROTECTED] wrote:
The information (or lack of it) on upgrading from zaptel to that
@*^QW%^%!!! dahdi is very frustrating.
I cannot find anything on how to uninstall zaptel, i found an earlier post
The information (or lack of it) on upgrading from zaptel to that
@*^QW%^%!!! dahdi is very frustrating.
I cannot find anything on how to uninstall zaptel, i found an earlier post
to this list which suggested make uninstall and make remove in the zaptel
directory which just generates errors
Is there an install script or step-by-step instruction somewhere on
whaty is needed to migrate from zaptel to dahdi?
I read the document that digium published which nicely states some of
the differences between zaptel and dahdi but i was looking more for
something like step-by-step
On Wed, 17 Sep 2008, Jared Smith wrote:
On Wed, 2008-09-17 at 19:58 +0200, Remco Barendse wrote:
Why doesn't Asterisk allow both usernamepass as well as setting an ip
adress on a sip.extension?
It does. To enforce ACLs on a SIP user or peer or friend, simply use
permit and deny statements
Maybe a bit silly question, but why doesn't Asterisk accept if you set
both a usernamepassword as well as an ip address for a phone?
My fixed phones in my home all have a fixed ip address, but i also have 2
Nokia GSM phones that can talk sip wich i would like to use from public
wifi.
It's
Hi list,
My caller ID is not working anymore on my TDM11B (TDM400P) cards and i get
this error message on the asterisk console:
== Starting post polarity CID detection on channel 4
-- Starting simple switch on 'Zap/4-1'
[Jul 8 11:58:55] WARNING[9539]: callerid.c:219 callerid_get_dtmf:
For quite a long time already my CallerID stopped working (maybe even when
i upgraded from Asterisk 1.2 to Asterisk 1.4). I am using a TDM400P card
(in TDM11B config) with one FXO and one FXS port.
Tried googling for some more recent examples of Asterisk config files for
use in The
On Mon, 7 Jul 2008, Matt Gibson wrote:
I think there is an issue with the screen refresh, mine also displays
searching... unless I reboot the phone, and leave wifi on when it boots
up, at this point it says internet calling: available .. but, it works
either way.
or maybe i am using an old
on, and calls route over wifi/voip when I am
registered instead of the cell network. Hth!
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barendse
Sent: Sunday, July 06, 2008
On Thu, 3 Jul 2008, Matt Gibson wrote:
http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/
Thanks for the link!
I installed and configured the phone according to the above link.
It only seems to work partly though. I
I just bought a HTC TyTn II phone, but unfortunately it doesn't even have
a SIP client in it.
I tried the wiki searching for a SIP or IAX client but only found some
PocketPC stuff (Windows Mobile 2003).
Does anyone know of a good quality SIP or IAX softphone that will run on
Windows Mobile 6?
I think the other guy would be. me ?
Unfortunately i am also running my asterisk on a production environment
where people start screaming the moment it doesn't work
I have 1.4.21 running at 3 locations in a home environment, simple TD400
cards with analog ports and no problems.
My problem
Ip upgraded yesterday from Asterisk 1.4.20.1 to 1.4.21
The update seems to work ok, when asterisk is started all is fine.
However after some time it is not possible to call anymore, my Snom
display simply shows Not available and incoming calls from the PRI fail,
like the PRI is not connected.
On Fri, 14 Mar 2008, Adrian Merwood wrote:
In my asterisk (Trixbox) server I would like to be able to dial
numbers from my address book using HUD or the SIP client on my 3G
phone using numbers in this format.
On asterisk I would like to strip of the + and replace it with an
international
Hi all!!
I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk
1.4.18. Both are home PBX's and both boxes register to a SIP DID at
exactly same provider. One box runs without errors on the console, the
other box keeps repeating :
[Feb 11 23:40:29] WARNING[11292]:
You can use the D option with the Dial command.
Something like this should work:
exten = _06,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})
It worked
Here is how i did it in FreePBX :
1) Setup a SIP extension for the ATA device, in my case i give it
extension number 298. Edit the
On Fri, 4 Jan 2008, EdPimentl wrote:
Have you looked into
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
-E
Yes i did, looks like an excellent product with many, many features and
of outstanding quality.
However, given the cost of that unit i would
I have an analog GSM Gateway that is connected to a normal SIP ATA device.
Basically what it does is this : when you call the extension nr. of the
SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia
On Thu, 3 Jan 2008, Benchev wrote:
Basically Grandstream HT286 is a single port FXS ATA.
In order to interconnect GSM gateway one would need FXO.
Are you sure it gives you new dialing tone or this is the * itself
you hear?
Yes, i am positive that i get a new dialtone from the GSM Gateway.
After upgrading from 1.2.25 to 1.2.26 i noticed that IAX - IAX calls
always result in Asterisk just exiting without any message.
Asterisk also seems to die when using a TDM400 with 2 FXO modules, placing
2 outgoing calls on both lines as Zap/g2 and then trying to make a 3rd
call.
Went back to
I wonder if there are any major obstacles for upgrading.
Just tried an in-place upgrade on my home box :
make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so
res_config_mysql.so; do /usr/bin/install -c -m 755 $x
I wonder if there are any major obstacles for upgrading.
My reasons for not moving to 1.4 :
- fear of possible instability problems, my 1.2 servers are rock solid
- fear of goofing up with the new way you have to configure asterisk
at install time (tell it which modules to build or not
On Fri, 26 Oct 2007, Benny Amorsen wrote:
RB == Remco Barendse [EMAIL PROTECTED] writes:
RB Hi list! Is anyone using the Kirk IP600/3 with SIP firmware
RB connected to Asterisk?
Yes.
RB If anyone would be willing to share the dump of their IP600 config
RB file, i would really appreciate
Hi list!
Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk?
Any experiences / caveats?
If anyone would be willing to share the dump of their IP600 config file,
i would really appreciate it.
Is there anything special i should put in my asterisk config?
Thanks !!!
Remco
Has anyone ever tried using a Nokia phone with SIP client as channel for
Asterisk? I mean i would like to receive calls to the mobile on
asterisk and use the Nokia phone to place calls to cell destinations.
I have enough Nokia E60's to do that and it would circumvent the need for
Is there really no way to have asterisk re-resolve domain names from iax
or sip providers if this failed or timed out the first time?
When asterisk boots on every box i have asterisk is t impatient
trrying to resolve the domain names for a first time. This results in
asterisk thinking the
Hi,
Greetings to All,
Im looking for some help on configuring VPN on the Asterisk PBX that I
have hosted in US. Im currently in Middle East and as everyone knows
some countries here has taboo to VOIP. Im not able to get phy phones
registered to my PBX as they are blocking SIP and IAX2.
Hi!
Is it just me or do the last 2 or 3 versions of the zaptel-1.2 branch seem
to break cli? Often not the full number is displayed, or only 2 or 3
digits?
I am in The Netherlands, and have had this in my zapata.conf (which used
to work flawlessly) :
signalling=fxs_ks
immediate=yes
On Mon, 16 Apr 2007, Martin Joseph wrote:
Just a warning for you all that are using Nokia series E phones for SIP
function.
I updated my phones firmware today using the Nokia Updater, and now the SIP
functionality, which previously worked pretty well is completely broken.
The phone no
On Thu, 29 Mar 2007, Carlos Jerónimo wrote:
Ive installed asterisk and freepbx. Through the interface ive
configured 2 extensions, 6000 and 6001.
My problem is that when i try to call from extension 6000 to 6001, i
hear this msg Im-sorryan-error-has-occured and the call is
terminated.
As
On Sun, 11 Feb 2007, Leo Ann Boon wrote:
Matt wrote:
I guess the question is... is it even possible to have a real-time VoIP
card running on PCIe? Or with 1,000 Interrupts a second.. does it simply
need to have its own IRQ?
Have you tried the Sangoma PCIe cards?
APIC is supposed to
I am trying to reconfigure an asterisk box that was using an HFC-S card
with bristuff but is now using 2 analog lines therefore I want to use the
TDM02B to connect to two POTS lines. The TDM02B has 2 red modules.
I have this in /etc/zaptel.conf
loadzone=nl
defaultzone=nl
fxsks=1-2
I have
Aarghh nevermind, my bad
A stock TDM02B comes with modules installed in slot 3 and 4, not 1 and 2.
For whoever might have the same problem and finds this post change below
to read:
fxsks=3-4
channel=3-4
On Sun, 11 Feb 2007, Remco Barendse wrote:
I am trying to reconfigure an asterisk box
On Sat, 10 Feb 2007, Matt wrote:
Hi folks.. just a few weeks ago I wrote this to someone else:
We have several 2900s in production as VoIP servers.. no lockups.
On every server I go into the BIOS and:
* Disable USB
* Disabled uneeded things like Parallel, Serial
* Put
On Sat, 10 Feb 2007, Andres wrote:
try booting with APIC and ACPI disabled?
Thats right. I have never seen a shared IRQ with Dell servers using APIC. A
RHL ES3 by default enables APIC so I have never even had to fiddle around
with it.
Ofcourse you don't. But simply because APIC makes
On Mon, 5 Feb 2007, Matt wrote:
We have several 2900s in production as VoIP servers.. no lockups.
On every server I go into the BIOS and:
* Disable USB
* Disabled uneeded things like Parallel, Serial
* Put ETH0 on a seperate IRQ from the Digium card
And everything's fine. Dell's do NOT have
On Sat, 3 Feb 2007, Gordon Henderson wrote:
On Fri, 2 Feb 2007, Remco Barendse wrote:
Would you be willing to share your blacklist for the kernel modules?
Have you considered compiling a custom kernel for your hardware rather than
not loading modules? It's something I've always done from
On Thu, 1 Feb 2007, Christophorus Laube wrote:
We have a 2850 in a productive environment with a BNE1 performing well
(OpenSuSE 10) and a 2950 with BNE1 and BN8S0 also performing OK (on Ubuntu
Edgy). You only have to blacklist some hotplug kernel modules and yes, we do
have very long pings (1
On Thu, 1 Feb 2007, Eric Rousse wrote:
Hi,
I was planning on getting a Dell PowerEdge 2950 for our new Asterisk
configuration.
But while searching for documentation about it and/or reported issues, I
found this:
http://www.voip-info.org/wiki/view/Asterisk+hardware
WARNING - many Dell
On Fri, 29 Dec 2006, Julian J. M. wrote:
It's not necessary to recompile the kernel for mISDN support. Check
http://www.laimbock.com/asterisk/
Grab the mISDN source rpm, and build it.
$ wget
http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm
$ rpmbuild
On Wed, 3 Jan 2007, Tzafrir Cohen wrote:
P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS
EXPERIMENTAL!)
..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown
signalling method 'bri_cpe_ptmp'
our Asterisk is not bristuffed. And you don't expect to use
Hi!
I noticed when upgrading asterisk that the latest version of asterisk is
not recognizing the version of asterisk-addons properly.
When you clean out /usr/lib/asterisk/modules and then install
zaptel-1.2.12 - libpri-1.2.4 - asterisk 1.2.14 - asterisk-addons-1.2.5
and then you compile and
On Fri, 29 Dec 2006, Julian J. M. wrote:
It's not necessary to recompile the kernel for mISDN support. Check
http://www.laimbock.com/asterisk/
Grab the mISDN source rpm, and build it.
$ wget
http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm
$ rpmbuild
Hi list!
I'm totally fed up with bristuff (or it's instability with a simple HFC-S
card), 2 out of 3 times when people try to call they get the information
tone that the number is not connected.
I would like to try vzaphfc and I am looking for information on it.
From previous posts I found
On Thu, 28 Dec 2006, Gavin Hamill wrote:
On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote:
vzaphfc is not a complete replacement of bristuff. It replies on most of
it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI
driver for HFC-s-based PCI cards.
Further, if
On Thu, 28 Dec 2006, Michiel van Baak wrote:
When you found out stuff, specially how to make stuff with a
simple HFC-S card stable please let me know.
We are not deploying them cards anymore because we never get
it stable.
Real simple setups can be done with a FRITZ!PCI card, but I
really
Hi list!
I have to do a new bare metal installation of a box running Asterisk with
bristuff or vzaphfc.
The box will be used as a really lightly loaded file server and pbx.
Any advise on which architecture I should use? The cpu is a 64 bit capable
AMD (the box is running x86_64 now) but is
I am using Asterisk 1.2.13 with Zaptel 1.2.11, I used to have an old PBX
connected to one port and the PRI connected to the other.
I'm having serious stability issues with Asterisk on a box that has been
rock solid previously.
The old PBX died two months ago so one port on the TE210P is now
BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that
the only place from which you can download an up-to-date version
nowadays is the Debian zaptel package:
http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/
http://packages.debian.org/zaptel-source
Thanks! I
After a reboot, asterisk is usually too much in a hurry to try and resolve
my iax/sip providers.
Asterisk starts before the internet connection is up and dns is working.
Then asterisk just waits, and waits and waits and waits even longer before
ever trying to revolve any voip provider again.
On Mon, 23 Oct 2006, Andreas Sikkema wrote:
Remco,
Asterisk starts before the internet connection is up and dns
is working.
knip
And then people say nightly asterisk restarts are not a good idea
Why is your asterisk startup script running before networking has been
On Sat, 21 Oct 2006, Michiel van Baak wrote:
On 20:15, Sat 21 Oct 06, Tzafrir Cohen wrote:
Interesting. Latest bristuff chenges the default Zaptel echo canceller
to MG2 (which is also the recommendation of Digium now).
BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It
On Fri, 20 Oct 2006, Michiel van Baak wrote:
On 02:39, Fri 20 Oct 06, Tzafrir Cohen wrote:
On Thu, Oct 19, 2006 at 11:27:07PM +0200, Michiel van Baak wrote:
On 23:04, Thu 19 Oct 06, Vidar wrote:
Bristuff has been updated;
On Sat, 21 Oct 2006, Michiel van Baak wrote:
On 09:39, Sat 21 Oct 06, Remco Barendse wrote:
And ofcourse half of the modules from this release do not build on a
x86_64 box :(
What is it that you use in bristuffed that is not in plain
asterisk ? I found myself battling with bristuff all
And ofcourse half of the modules from this release do not build on a
x86_64 box :(
Could you please be more spesific?
What distribution? What kernel? What errors?
Found the problem, it was fairly limited, where all the digium
stuff finds the kernel automagically, the bristuff things
Ah, you are using it for the hfc-pci cards. That's a valid
reason ;)
What's the latest BRISTUFF version that does work on x86_64
for you ? One of our customers reported trouble with this
bristuff on x86_64 as well and I dont have a _64 machine to
test.
The last working release is the one
On Sun, 15 Oct 2006, Les Bell wrote:
Cutting to the chase: I'm not aware of any audio problems, but our system
doesn't get heavy use (only two lines and eight phones).
OK, thanks for the reply.
The anouncement at trixbox.org is not very clear on this. There is
reference to 'distorted voice
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