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
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
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
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
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.
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
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?
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
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
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
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:
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
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
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
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
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
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
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
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.
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 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 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 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, 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 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 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
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.
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
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
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
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 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.
___
--
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 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
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 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
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
If memory serves correctly, most of the above has been raised as issues in the
past and the suggested work around has been to run a dns caching server on the
asterisk box.
FWIW, I always use IP addresses instead of dns names. But, I don't have to
deal with dynamic ip changes of any device
On 9/2/06, Daniel Pocock [EMAIL PROTECTED] wrote:
http://www.readytechnology.co.uk/open/ipp-codecs
Asterisk 1.2 support coming shortly.
Asterisk 1.2 support? I'm using your codecs ever since 1.2 was released?
Even though Asterisk always complains about modules being present that may
Same problem here on CentOS 4.4 :(
Strange that apparently the tarball was not tested if it would even
compile
On Fri, 8 Sep 2006, Stuart Sheldon wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I can confirm the same problem, it looks like the oct612x directory tree
is missing from
Hi list!
I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the
TEI check request message were I was getting errors.
Concerned about that I switched to plain vanilla bristuff.
Now everything *seems* to be working without errors but I regulary get
reports from people trying
Hi list!
Does anyone have experiences with the updated model of the Kirk IP600?
The V3 model is supposed to support SIP instead of only SCCP or H323 which
would make the use with Asterisk a lot easier.
I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is
still giving me
On Thu, 14 Sep 2006, Henrik Woffinden wrote:
Hi
Right now I'm running Asterisk with ZapHFC BRIstuff and it work, but
with some MSN addressing problems on the ISDN bus. I've had no success
solving the problem. Problem is making 3 MSNs ring on one B-channel.
I thought of trying mISDN
On Fri, 22 Sep 2006, Conrad Wood wrote:
On Mon, 2006-09-11 at 21:14 +0200, Remco Barendse wrote:
Hi list!
I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the
TEI check request message were I was getting errors.
Concerned about that I switched to plain
On Fri, 22 Sep 2006, Michiel van Baak wrote:
On 18:22, Fri 22 Sep 06, Remco Barendse wrote:
It seems that development on bristuff is stalling a bit, maybe because
Asterisk is working on native support of MISDN.
Hmm,
Will the quad/octobri and gsm cards be supported by MISDN ?
I think
Thanks for this, I was looking for this too.
Will the DTMF tone be audible to the other side? (In other words will they
know something is happening)
On Thu, 5 Oct 2006, Joel Hill wrote:
Hi Noro,
Depending on what firmware you have this is the way to go.
Go to the Functions keys page, then
On Thu, 5 Oct 2006, Joel Hill wrote:
No worries. Good question, I wasn't sure so I just tested it and it seems that
the answer is yes it does send the tones to the other side.
Can I ask why this would matter, I think there could be legal implications of
recording a call and not notifying the
Remco Barendse wrote:
Thanks for this, I was looking for this too.
Will the DTMF tone be audible to the other side? (In other words will they
know something is happening)
On Thu, 5 Oct 2006, Joel Hill wrote:
Hi Noro,
Depending on what firmware you have this is the way
Hi list!
I recently upgraded to FreePBX 2.1.3 although I am not sure if this has
something to do with it.
I do a nightly restart of Asterisk, just in case. This has been working
fine months but since a few days asterisk seems to die and I am not able
to restart it again, I keep getting a
On Wed, 11 Oct 2006, Tzafrir Cohen wrote:
On Wed, Oct 11, 2006 at 09:14:55AM +0200, Remco Barendse wrote:
Hi list!
I recently upgraded to FreePBX 2.1.3 although I am not sure if this has
something to do with it.
I do a nightly restart of Asterisk, just in case.
Why?
Sometimes
Sometimes the internet connection is dropped and asterisk doesn't do a dns
lookup and provider re-rest quickly enough so all calls are going out via
expensive ISDN.
So detect a connection change and then restart, by the way of 'asterisk
-rx restart now' (or 'restart when convinient',
On Thu, 12 Oct 2006, Eric ManxPower Wieling wrote:
Matt Florell wrote:
If you downgrade, let us know if it fixes things for you.
It's strange that there were so many changes in the 1.2 SVN branch
after 1.2.7.1 that seem to be complete changes in how some things
operate(like the
Hi list!
I read the trixbox anouncement that they downgraded to kernel version 34
to get rid of some audio problems
I'm not running trixbox but normal Centos 4 with asterisk installed. I
tried to find some further info on this but couldn't find any.
Do audio problems occur with normal Centos
I'm now running kernel-2.6.9-42.0.3.EL
Not really an answer to your question, but I found out all kernels above
2.6.16 do a better job on asterisk systems then the ones before that. No idea
how this is possible as I'm in no way familiar with the inner workings of the
linux kernel, but I
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
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 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
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 Wed, 31 May 2006, whois wes wrote:
Sorry I'm late, just joined the list.
We have 400 seats spread across 8 Dell 2805's, and were running TE410P's in
all of them. After six months of troubleshooting call drops, static, and
random server lockups, we moved 4 of the servers to Sangoma A104d's
On Wed, 31 May 2006, Steven wrote:
What were the kernel parameters that you changed? (what OS, by the way?)
I am running CentOS 4.3, but have not changed any kernel settings yet.
Nothing exciting, just adding noapic did improve a lot on the hits:
title CentOS (2.6.9-34.ELsmp)
root
Hi list!
I have a TDM11B (DevKit). When I start asterisk everyting is working fine.
However as soon as I do a RELOAD from the console the FXS port stops
working. I have a phone connected to it, when I pick up the phone after
issuing a RELOAD I get this on the console:
Jun 1 20:27:38
Hi list!
Does anyone have a small tarball with some example config files for Snom
mass updating under linux? The Snom docs are not very clear to me and
they nicely specify some DHCP parameter numbers(??) and how to do that
when running windoze servers but the linux info is very thin.
Hi list!
Are there any changes in the behaviour of the Dial command between
1.2.7.1 and 1.2.8.?
I am forwarding calls to my legacy PBX using :
exten = s,1,Dial(Zap/g1/8210,90,r)
Ever since I upgraded to 1.2.9 it seems as if the Legacy PBX is no longer
receiving the extension I am calling on
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except
for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could
On Tue, 6 Jun 2006, Michiel van Baak wrote:
On 15:46, Tue 06 Jun 06, Remco Barendse wrote:
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1
On Tue, 6 Jun 2006, Michiel van Baak wrote:
On 15:46, Tue 06 Jun 06, Remco Barendse wrote:
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except
for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could
It may seem like overkill, but I have really tried anything to get fax
reliable, without success.
Instead of a channel bank you could also use an old legacy pbx for that. I
am using our old pbx just for faxing, doing bridging on a TE210P. A E1
channel bank will cost a fortune and if your PRI
Hi list!
I have a Centos 4.3 box running Asterisk 1.2.9.1 with FreePBX 2.0.1
I noticed that this setup is keeping a full asterisk log which, after 1
month in production, has already grown to 1300 Mb in size. This is the log
location : /var/log/asterisk/full
Why is this on by default (I
Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a
x86_64 box (I guess nobody is using x86_64 platform or is able to fix this
themselves?)
First of all when bristuff is downloaded and compile is started it appears
that the bristuff Makefiles are badly broken.
The
found it, in bristuff-0.3.0-PRE-1q/zaphfc/Makefile
again it is required to change KSRC=/usr/src/linux/ to
KSRC=/usr/src/linux-2.6/
I wonder why neither florz nor kapejod fixes these problems (several
modules do not compile).
I will not try running bristuff anymore without florz but from
I'm running bristuff-0.3.0-PRE-1q
The line seems to work but I get these messages on the screen every few
minutes :
1 received TEI check request for TEI = 81
Jun 20 20:18:05 WARNING[27571]: chan_zap.c:8503 zt_pri_error: 1 TEI remove
TEI = 127
Jun 20 20:18:05 WARNING[27571]: chan_zap.c:8503
I tried compiling Asterisk 1.20 (fresh cvs checkout of just some 15 minutes
ago) on a CentOS 4.2 box.
Compiling zaptel seems to work fine but when I try to compile * I get this:
chan_zap.c:8904: error: structure has no member named `useruserinfo'
chan_zap.c:8012: warning: unused variable
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
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
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
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!
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
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
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
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