Hi all,
Having trouble registering Linksys SPA941 with our * box. We can't
find an entry in the config screens to allow us to put the IP address
of the SIP server (i.e. the * box) in.
We can find an entry for a SIP proxy in the phones set up, but we're
not using one (SIP connections are
if this is what is causing your issues or not, but hopefully it will
help if it is.
Cheers,
Damian Funnell.
FFF Managed Technology Ltd.
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz
Quoting Health Masters [EMAIL PROTECTED]:
We
Polycom make some surprisingly good and reasonably priced SIP handsets
too. Many of these support headsets and we've been quite impressed by
them.
FFF Managed Technology Ltd.
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz
Hi,
Does anyone have remote call pick-up working on * (either via SIP or
otherwise)? If so then can you post your features.conf, sip.conf and/or
zapata.conf?
We can't seem to get this (seemingly simple) function to work.
Cheers,
Damian.
___
Hi,
Does anyone have remote call pick-up working on * (either via SIP or
otherwise)? If so then can you post your features.conf, sip.conf and/or
zapata.conf?
We can't seem to get this (seemingly simple) function to work.
Cheers,
Damian.
--
FFF Managed Technology Ltd
60 Cook St
P.O. 6368
Hi Andrew,
Not sure if I understand your question, but this may help - * has the
following settings in features.conf that are related to parking:
parkext = ;the extension that users xfer calls to in order to
park them
parkpos = - ;the extension range that * will use to park
Have you checked that the TDM400P isn't sharing an IRQ with anything
else? Don't trust /proc/interrupts - run lspci -v to confirm this.
We have * running on an x206 and found that the only way to stop the
TDP400P sharing an IRQ with other devices was to juggle cards between slots.
Hope this
Hi,
Has anyone got sample sip.conf, features.conf and zapata.conf files that
they can send me that demonstrate a working remote pick-up config? We
can't seem to get it working at one of our sites - have changed the
remote pick-up extension so it doesn't conflict with the SIP phones
redial
,
Waldo
On May 16, 2005, at 12:59 AM, Damian Funnell wrote:
Hi Waldo, it really depends on who you ask - Digium say that
anything less than 99.99% is going to result in problems, but ours
regularly runs at around 99.98% and we don't have any problems.
One of our boxes was running at around 99.96
Hi Rich,
This is always a BIOS setting - there is no O/S command to disable H/T.
To date I have never heard of a BIOS that does not allow the user to
disable H/T, but I have read that there are BIOS'es out there that don't
offer this function.
Go into your BIOS setup screen and you should
...Jens makes a liar out of me, although I read that the 'noht' switch
stops the OS from using H/T but doesn't disable it completely. I make
no warranties regarding the accuracy of this information, though.
D.
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356
Rich, did you check IRQ's? Our zttest results didn't improve markedly
when we did either change (IRQ's or H/T), but the problem went away
regardless.
Other than that Digium have recommended throwing out our SCSI320 RAID
hardware and replacing it with IDE (i.e. not SATA) kit, although
Hi Gregory,
Have you checked that the card is on its own IRQ?
There has been a bit of discussion about this type of thing recently on
the list, do a search on the archive to find the various threads.
Try running zttest and see what accuracy it is reporting - anything less
than 99.99% is
Hi Waldo, it really depends on who you ask - Digium say that anything
less than 99.99% is going to result in problems, but ours regularly runs
at around 99.98% and we don't have any problems.
One of our boxes was running at around 99.96% and we had major issues
with the voice quality packing
Hi team,
Not long ago a bunch of us were posting reports of a strange phenomenon
where voice quality would pack up completely from time to time,
typically resulting in loud crackling on the line and/or the voice
channel breaking up completely. With our installation it would occur
from time to
Hi Tony, check out my recent post regarding our experiences with
Hyperthreading and * with Zaptel cards.
We have a few machines in the wild that *do* run Hyperthreading but no
Zaptel cards and these work absolutely fine. My understanding is that
the Hyperthreading problems are purely related
Hi all,
We've got a problem where a bunch of SNOM 190 phones that we have just installed
are giving us problems with DTMF tones.
Users of all phones reported that when they access voicemail the VM app is not
recognising DTMF tones. One clever user figured out that they DO work if you
hold the
Any takers? Sometimes the most basic questions yield the least replies,
huh?
Cheers,
Damian.
Original Message
Subject: Call forwarding
Date: Wed, 04 May 2005 08:40:41 +1200
From: Damian Funnell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi team,
Basic question I know, but I can't seem to find any obvious information
about this:
Does anyone know if * natively supports call forwarding from a given
extension (i.e. call forwarding without having to write a macro)?
My user wants to be able to dial a code plus a phone number to
Hi all,
Does anyone know what the easiest way is to delete voicemail for one
extension? Had a search online but couldn't find anything.
Cheers,
Damian.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I agree - you guys really shouldn't be wasting our bandwidth unless it's
important.
What were you thinking?
trixter http://www.0xdecafbad.com wrote:
On Wed, 2005-04-20 at 22:14 -0500, Dan Perik wrote:
Michael D Schelin wrote:
Ok you guys enough. The debate will go on forever.
Hi Tim,
Thanks for your post, it's most insightful. It certainly puts a pretty
large dent in my confidence in the TDM for commercial use - imagine if
there was more than one TDM in a system (especially with a RAID adapter).
Running a PABX without hardware RAID 0 is not an option for us, as we
Hi everyone,
For anyone who was following the earlier thread about noise problems on
*, here's a little gem from Digium support.
In a nutshell Digium told us that they thought that low accuracy results
from zttest were the cause of noise problems that we have been
experiencing. We
Hi Joseph,
Let me take a guess - the problem only occurs when dialling four digit
extensions?
I think you will find that your dial plan is matching the three digit
extension and then dialling it straight away - Asterisk won't wait for a
timeout before trying to follow the dial plan, as soon as
.
Bryan Boatright wrote:
Is the APIC and IO-APIC enabled? Send us 'cat /proc/interrupts' and
your /var/log/boot.msg (or your distro's equivalent bootup log).
Damian Funnell wrote:
Hi all,
I've found that a TDM400P card in our * box is sharing IRQ's with two
other
Hi,
For those that were having the same line noise problem that we were, an
update:
* Our TDM400P *was* sharing an IRQ, despite the output from 'cat
/proc/interrupts' showing that it wasn't. Running 'lspci -v'
showed that it was and we had to perform some card juggling to get
The following dial string dials both extensions - this has worked for
SIP and analogue extensions on our Asterisk machines. Both extensions
ring until one is answered:
exten = s,3,Dial(SIP/9295SIP/9287,,t,)
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
Rotate or make sure the line you are dialling out on isn't in use before
you try and use it? Lookup setgroup/checkgroup on the Wiki if it's the
latter -
http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup
Allows you to create logical groups for just about anything, check
whether the groups
Digium have told us that a problem that we are having (with accuracy of
zap interface as measured using zttest) may be due to the fact that we
have a Xeon processor with hyperthreading and have suggested turning H/T
off.
Anyone else experienced a problem like this? No too keen about turning
Hi Kib,
What exactly is it that you want to do? If you have a direct dial-in
(DDI) number that goes to a certain extension then you can handle this
pretty easily in your dial plan - check out this snippet below from one
of our customers' machines. This example is pretty basic, but it works
Hi Robson,
We are using a 4 port Eicon Diva Pro BRI ISDN with a TDM400P with four
FXO ports. We are using CAPI with the TDM400P and everything works fine
most of the time.
We are having periodic problems where the call quality completely falls
apart for all in-progress calls and we have yet
Hi all,
I've found that a TDM400P card in our * box is sharing IRQ's with two other
devices. The server doesn't support assigning IRQ's through the BIOS and the
pig only has three PCI slots, so swapping cards between slots hasn't fixed the
problem (it just ends up sharing IRQ's with other
Hi Rich,
Hear your point about the trace and all, will try and figure something
out. Will also look at logging debug messages.
We did the unthinkable and purchased a support incident through Digium
and they have zeroed in on the zttest output, as per the info below
(I've pasted in an excerpt
Hi Manish,
Sure can, although you will need a timing source. If you don't plan to
have any Digium hardware then you can use ztdummy (see
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy), which we have
never used personally, but have yet to hear bad things about.
D.
FFF Managed
Hi Ronald,
We use SetGroup/CheckGroup and your syntax appears to be fine (either
that or ours is broken too, but it seems to work ok!)
One question - what lines do you have in priority 1 - 3? We found that
our dial plan would not work unless the first priority (for all
extensions) was 1 and
to zero - we have a
total of four SetGroups and these are set and checked extensively
through our dial plan and they work fine.
Ronald Wiplinger wrote:
Damian Funnell wrote:
Hi Ronald,
We use SetGroup/CheckGroup and your syntax appears to be fine (either
that or ours is broken too, but it seems
Hi Paul, there was a thread
yesterday in regards to a few of us experiencing a very similar problem
- a problem that (if the same for all of us), doesn't seem to have been
properly diagnosed yet.
One thing that appeared to be common to all of us was the version of
Asterisk that we are running
phone and test it to see if its the phone. Let me know if you come
up
with any ideas.
Paul
From: Damian Funnell
[mailto:[EMAIL PROTECTED]]
Sent: Monday, April 11, 2005 12:49
To: [EMAIL PROTECTED];
Asterisk Users Mailing List -
Non-Commercial
!!!
- Andre
-Original Message-
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
*Damian Funnell
*Sent:* Monday, April 11, 2005 3:08 PM
*To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
*Subject:* Re: [Asterisk-Users
Thanks for that Rich. Etheral trace is going to be almost impossible
for various reasons, but will try the other two options.
Can't find much online re. debugging - any chance of killing the box by
turning this on?
SIP show channels and the various CAPI show commands do not show
anything out
Rich Adamson - would appreciate your advice as well, as your mail is the
closest I have seen to a knowledgeable response so far in regards to
this crackling issue. I have a customer who has a very similar
crackling problem and to date we have suspected it to be the ISDN BRI
adapter and/or
I have a very similar problem that
I have been grappling with for a while. I've got a genuine TDM400P
with four FXS ports and am using an Eicon Server quad BRI ISDN (using
CAPI) for external calls.
To date we have had no luck at all in diagnosing this problem as we too
have periodic problems
Forgot to mention - we are using
an IBM xSeries 206 Server, so the Dell riser card may not be the issue
if we are having the same problem.
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz
Damian Funnell
Hi Courtney,
This really depends on the type of channel you are using. We use CAPI
with one of our customers and they have ten DID's (DDI's in NZ) and use
four BRI lines. Here is a snippet from their extensions.conf that
shows how it dials certain extensions based on the number dialled. The
Hi there,
We recently did our first * install with CAPI and we found the levels
of support (and general knowledge) within the community seriously
wanting. In fact, we found things so bad that I would caution against
using CAPI unless you are feeling particularly game and confident in
your
suspected problem was to do with BT102's (i.e. SIP
handsets), but problem has since occurred with analogue phones as
well.
Appreciate any help with this, as source of problem not immediately
apparent and customer patience wearing pretty thin.
Cheers,
Damian Funnell
Hi all,
I have two questions regarding CAPI. Excuse the fact that they are
very 'newbie' in nature, but the CAPI documentation is wafer thin!
Firstly I have four BRI adapters (all trunks and controlled by CAPI) in
my * box and I would like to know whether I can group these together
for
Thanks Elmar. I assume it is up
to the carrier to determine the MSN for each connection?
D.
FFF Managed Technology Ltd
60 Cook St
P.O. 6368 Wellesley St
Auckland
t +64 9 356 2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz
Elmar Haneke wrote:
Lastly, my capi.conf (as below) only
Hi all,
Can anyone help me with a CAPI problem that I am having. I've got one
BRI trunk (will have 4 when it goes into production) and when one of
the B channels is in use (i.e. there is an incoming/outgoing call in
progress) I can't get Asterisk to answer the other ringing B channel
Hi Ryan,
I've used the BudgetTone 101 on several accounts and they certainly
aren't the best phone on the market, but they have so far been reliable
(touch wood) and are pretty straightforward to set up.
Call quality would probably rate at a 8 out of 10 on these phones, but
that's not much
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