Hi,
Is there any way to get asterisk to be able to use USB devices like the
AUP-03 shown on this website?
http://www.chronos.com.tw/products/usb/Skype/skypephone.htm
Thanks,
Zoltan
--
==
Geograph (Pty) Ltd
P.O. Box 31255
Tokai 7966
South Africa
Tel:+27-21
Hi,
Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per
http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone
The dialplan was configured through AMP and has nothing fancy in it.
As a first time user of not only Asterisk, but also a PBX, there are
some operator teething prob
David Burgess wrote:
Matt Riddell wrote:
David Burgess wrote:
Hi,
I am new to the list.
I have just *re*-installed and rebuilt asterisk from the head
branch and I am left with the problem that the sound bounces
around. When installing zaptel I get the following message from
ztcfg.
Las
Tzafrir Cohen wrote:
Hmmm... didn't you say you had a x100p card somewhere on your system?
Nope - only the 2 pci HFC ISDN cards.
OK - I hope I'm not buggering up the thread, but I need to answer myself
as there now seems to be only 2 remaining concerns - but then I haven't
yet had a chance
e ISDN cards are seen & ready for use ???
Please could someone check my hybrid zapata.conf at the end of this
email ???
Cheers & thanks,
Zoltan.
Zoltan Szecsei wrote:
Hi All,
It's broken !! (drat)
Asterisk if failing to load with the following error (taken from end
of /var/log/a
Hi All,
It's broken !! (drat)
Asterisk if failing to load with the following error (taken from end of
/var/log/asterisk/full) after adding bristuff.
Can anyone help please?
Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone'
(Standard Linux Telephony API Driver)
Jul 17 19:57
Tzafrir Cohen wrote:
On Sat, Jul 16, 2005 at 07:28:06PM +0200, Zoltan Szecsei wrote:
Tzafrir Cohen wrote:
On Sat, Jul 16, 2005 at 05:40:28PM +0200, Zoltan Szecsei wrote:
Tzafrir Cohen wrote:
Don't know about [EMAIL PROTECTED], but upcoming version of Rapi
Rudolf Ladyzhenskii wrote:
Hi, all
I have couple of SIP phones and they are in [from-sip] context.
I also have an IAX2 phone. I have put this one in [iax-user] context.
I want to make calls between SIP and IAX2 phones. If I put them all in
same context all is fine, however when they are in di
Carl Andersson wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan
Szecsei
Sent: zaterdag 16 juli 2005 12:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk
Hi,
Can anyone please point me in a direction as to how to set up these 2
pci cards with AAH 1.1?
I have (am still) googling left, right & center - but haven't found a
definitive guide yet.
The centos based setup lacks any of the tools I know (insmod, modprobe
) so it is time consuming
Carl Andersson wrote:
Maybe this is rather a hardware question, but I am posting it on this
list because the probability of someone else of you having tried this
is greater here than other places I can think of.
I have an ISDN card that is setup in NT mode using the zaphfc driver
in bristuff
Dave Cotton wrote:
On Thu, 2005-07-14 at 16:31 +0200, Zoltan Szecsei wrote:
3) The alias suggestion
I did not understand this at the time I received it - Had I noticed
Tzafrir's pointer to the ethernet HOWTO, I would have realised that
alias in this context was not giving an ali
Zoa wrote:
Could someone also tell me how much a car costs ?
What i mean is, it all depends on your server and the codecs used, the
max is currently a DS3 worth of calls.
Ah - glad you clarified - I thought it depended on the make & model of
car you wanted
:-)
Z
altus wrote:
Good da
Tzafrir Cohen wrote:
If you want useful answers from people here, provide some data for
people to work with. As for that data: people asked you to look at some
specific files.
E.g: frankly I still can't tell if both cards get loaded by the same
module or by different modules. Frankly, I can't t
Dave Cotton wrote:
On Thu, 2005-07-14 at 09:01 +0200, Zoltan Szecsei wrote:
Ken Godee wrote:
On reboot sometimes my onboard gigabit nic gets eth0 and sometimes
the pci 3COM gets eth0 and this causes havoc with another piece of SW
I run.
I seem to remember having this type
James Oakley wrote:
On Wednesday 13 July 2005 12:40 pm, Zoltan Szecsei wrote:
Hi All,
Long time no chat ;-)
Asterisk 1.0.9 (sometimes) won't authenticate IAX phones after re-boot
of SuSE 9.3 box
I've traced the problem to be with the firewall and the fact that I have
2 NICs
Ken Godee wrote:
On reboot sometimes my onboard gigabit nic gets eth0 and sometimes
the pci 3COM gets eth0 and this causes havoc with another piece of SW
I run.
Is it actually "ethx" getting flipped or the ip addresses?
___
Asterisk-Users mailin
Kevin P. Fleming wrote:
Zoltan Szecsei wrote:
I've traced the problem to be with the firewall and the fact that I
have 2 NICs in the box. Now that I have opened port 4569 on both
interfaces, asterisk seems happy *but* does anyone know how to force
SuSE 9.3 to always bring up a specifi
Hi All,
Long time no chat ;-)
Asterisk 1.0.9 (sometimes) won't authenticate IAX phones after re-boot
of SuSE 9.3 box
I've traced the problem to be with the firewall and the fact that I have
2 NICs in the box. Now that I have opened port 4569 on both interfaces,
asterisk seems happy *but* doe
Rich Adamson wrote:
If you don't mind, can you explain me a little more the ztcfg problem. My
experience with that is null, but I need to setup a box for testing purposes
only, and I don't want to but a TDM card only for make it work.
I know people that has asterisk running on SIP that don't us
Rich Adamson wrote:
If you don't mind, can you explain me a little more the ztcfg problem. My
experience with that is null, but I need to setup a box for testing purposes
only, and I don't want to but a TDM card only for make it work.
I know people that has asterisk running on SIP that don't us
Hi Carlos,
OK - I speak from memory and a little bit of newbie fiddling (which
thanks to you and Rich took a successful turn).
Carlos Alperin wrote:
Zoltan,
If you don't mind, can you explain me a little more the ztcfg problem. My
experience with that is null, but I need to setup a box for
Carlos Alperin wrote:
Zoltan,
If you don't mind, can you explain me a little more the ztcfg problem. My
experience with that is null, but I need to setup a box for testing purposes
only, and I don't want to but a TDM card only for make it work.
I know people that has asterisk running on SIP th
Yeee-h!
doesn't this look pretty?
***
Asterisk Ready.
*CLI> -- Accepting AUTHENTICATED call from 192.168.0.201, requested
format = 4, actual format = 4
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "IAX2/z2|20|tr") in new stack
-- Called z2
Ha! yes - we are getting there - hopefully soon you will allow yourself
some time for anything other than me.
see inbetween - and then at the end.
Rich Adamson wrote:
Now we're getting there. In one of your previous emails, you indicated:
8) IAX username - still left blank
9) IAX password -
Rich Adamson wrote:
wassim darwish wrote:
how to edit the time of ring "3ms" to "4ms" in
astcc since it displays this on console "Nobody picked
up in 3 ms" when nobody picked up the phone in
3ms and then it hangup.
please help i have been asking this question from long
time
Tzafrir Cohen wrote:
On Fri, Jul 08, 2005 at 11:12:37PM +0200, Zoltan Szecsei wrote:
Is this how the modprobes are supposed to respond??
gl0:/home/zls # modprobe zaptel
gl0:/home/zls # lsmod | grep z
Module Size Used by
zaptel239620 0
crc_ccitt
Rich Adamson wrote:
Yes, the above indicates the phones did in fact register "at least
one time", as indicated by the IP address in the Host colume.
The Status = Unreachable is saying the phones no longer are
reachable by asterisk. So, either the phone is going to sleep
on its own, or, for wha
Carlos Alperin wrote:
Zoltan's problem looks more to be a timing issue, just he is not using
Zaptel but ztdummy modules that are not loading.
Regards,
Carlos
Is this how the modprobes are supposed to respond??
gl0:/home/zls # modprobe zaptel
gl0:/home/zls # lsmod | grep z
Module
Rich Adamson wrote:
Yes, the above indicates the phones did in fact register "at least
one time", as indicated by the IP address in the Host colume.
The Status = Unreachable is saying the phones no longer are
reachable by asterisk. So, either the phone is going to sleep
on its own, or, for wha
Rich Adamson wrote:
Well, if I were working with those I'd fire up Ethereal and look to
see "exactly" what the phone was doing. If they really are iax capable,
you should see at least some iax packets coming from them. Then, the
real answers to why the phone doesn't work can be resolved from the
Rich Adamson wrote:
I still think it's a ztdummy issue - is that also needed with SIP?
Zaptel timing (eg, ztdummy) isn't needed for iax or sip. Its only
needed on a couple of application items like the meetme, etc, which
are documented on the wiki.
You're quite right about SIP not n
Hi Rich,
See debug output after your post below.
Thanks,
Zoltan
Rich Adamson wrote:
"If" putting a host= statement in * does anything in terms of completing
the registration, there is a serious infrastructure issue. That isn't
going to lead him to resolving the registration problem.
Why not
Carlos Alperin wrote:
The timing is provided by Zaptel, if you don't have that there is no timing.
So, after we get the registration, the next step is get the TDMoE working,
if not you cannot generate any calls.
If I confuse you, I'm sorry but was trying to help not to make you loose.
I don't
Carlos Alperin wrote:
If I write try with sip, everybody will try to hang me in the highest tree.
:-)
(you have a nice attitude)
So, If was on your position, I'll try SIP.
I did actually have a "quick" try - and got similar problems - but I'll
have to have a good look to know if the
Ha! (again)
Hopefully this helps you guys a bit (to help me )
Ta yet again,
Zoltan
From the wiki page about my phone, I changed my iax.conf to:
(note 6 lines after CallerID)
* sip.conf **
[general]
port=4569
bindaddr=0.0.0.0
bandwidth=medium
disallow=LPC10
;den
Carlos Alperin wrote:
Ok,
Now you get registration, but still you cannot complete the call.
I don't see your dialplan on your files. However, the error log shows that
ztdummy is not working, so there is no timming.
Lsmod shows the module already installed?
Carlos
I did have it on a p
Ha!
I think I found my phone - see point 2 on my previous post (copied below).
Although the sticker on the bottom of my phone does have "ALDU", the
firmware version is 02.09.07
So, it should cope with IAX only (and no SIP at all), yet I cannot get
it registered (yet)
http://voip-info.org/tik
Rich Adamson wrote:
The iax2 show peers is indicating the phone is not registering
properly. (If the phone "never" changes IP addresses ever, then he
"could" put a host= statement in the iax.conf, but that is very
non-standard and will only serve to confuse people.)
In my reply to "time bandi
Time Bandit wrote:
*CLI> iax2 show peers
Name/UsernameHost Mask Port Status
z2 (Unspecified) (D) 255.255.255.255 0 Unmonitored
z1 (Unspecified) (D) 255.255.255.255 0 Unmonitored
From this, you can se
IAX2 SHOW PEERS and tell me what you get?
When you finish you can exit with quit.
I think that the system doesn't know where are the phones located.
Thanks,
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent: Friday, July 0
Carlos Alperin wrote:
The only reason for try SIP, is to find where the problem is.
You can use what you prefer, if you can made it works.
Any chance to see both SIP & IAX.conf?
Thanks,
Carlos
Hi Carlos,
Thanks for you help thusfar.
I have provided:
iax.conf
extensions.conf
bits of log
My opinion is to keep using IAX, because, like you concluded, it's a
better protocol.
hth
hth? - well, only if you can give me some pointers as to what I should
be looking at to make it work :-)
(all right, yes it does help: you've given me confidence in my
conviction, but not helped
Carlos Alperin wrote:
What about define those phones on the SIP.conf and use sip, instead of IAX.
That protocol use be more used to communicate Asterisk servers more than
phones.
Regards,
Carlos Alperin
Ah - ok - I understood from the docs that IAX was better and, as the
phone was capabl
Hi,
I'm trying to set up two ACT SIP/IAX capable phones to communicate with
each other on the same internal network, using asterisk 1.0.9 on SuSE
9.3 (because I intend to grow the situation after this basic setup is
functioning)
The phone IPs are set to 192.168.0.201 and 202 respectively.
I
Hi,
Sorry to re-post, but I'm still having hassles with ztdummy. I'm using
kernel 2.6.11.4-21.7-smp and Asterisk 1.0.8 on SuSE 9.3
The first 3 makes (see below) for zaptel work out ok - but the
ztdummy.ko (etc) files *are* created even though I haven't yet
uncommented ztdummy in Makefile.
Hi,
I'm following the instructions on Asterisk Doc Proj:
astersik_1.0.8/docs/docs-html_one/vm1.html#AEN30 and they don't seem to
work out.
The order suggested is:
cd zaptel
make clean
make linux26
make install
vi Makefile (and uncomment ztdummy)
make
modprobe zaptel
modprobe ztdummy
Th
Bob Goddard wrote:
On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote:
Hi Bob,
Thanks - I'll run with the README idea of yours.
Your comment regarding re-boot however is not valid. I also thought that
was the case and (as I said on the first line of my message) I
specifically reboote
pare the kernel for building thirparty modules.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan
Szecsei Sent: 01 July 2005 01:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] make error for zaptel
Hi,
I
Hi,
I'm running SuSE 9.3 fully updated using YOU and I *have* re-booted the
box (in a hope to sort out the uname -r issue mentioned below).
I'm using the Asterisk Doc Proj vol 1 to guide me through the initial
setup. I have no special HW and intend to use asterisk on an internal
network just t
Hi Hamish,
Sorry about being a day late... man, these lists are hell to keep up
with.
You could also try xlite - www.xten.com
Cheers,
Zoltan
Hamish Whittal wrote:
Hi Folks,
I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compil
Doug Lytle wrote:
Zoltan Szecsei wrote:
Hi,
Total newbie - I spent most of yesterday looking through various docs
and could not come up with anything useful.
I've searched for docs relating to how asterisk loads (and therefore
what makes it load certain modules) and couldn't fin
Hi,
Total newbie - I spent most of yesterday looking through various docs
and could not come up with anything useful.
I have 2 different SuSE 9.3 boxes - a P3 with 2 * ISDN BRI pci cards and
a P4 with no asterisk relevant HW at all.
I am using the default asterisk install that was loaded whe
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