Hi all,
Have a problem that I have been battling with for a few days now with
help from voiptalk.org support.but I thought someone here might have
seen this before.
I have an asterisk box running on a real non nat'ed ip address with an
incoming number from voiptalk.org on IAX2.
The problem I
Tony Mountifield wrote:
Mark Benson [EMAIL PROTECTED] wrote:
Hi all,
Have a problem that I have been battling with for a few days now with
help from voiptalk.org support.but I thought someone here might have
seen this before.
I have an asterisk box running on a real non nat'ed ip address
I have been using an IN1002 generic handset (supposed to be an unbranded
cisco copy but I am skeptical) for a few months (6months+) now, and it
seems pretty stable - however I haven't found a reliable supplier Also
there is almost no support for them..
I have switched to the grandstream
Cheers Tony,
That sorted it! - have passed this info onto voiptalk to update their
help pages
You wouldn't care to add an explanation as to why user works over friend?
Cheers,
Mark
Tony Mountifield wrote:
Mark Benson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Mark Benson [EMAIL
I am having problems transferring calls from one sip extension to
another - the extensions use various phones hardware/software.
From what I can tell I should just be able to press # and then dial an
extension to blind xfer a call right? How do I do attended xfer?
Either the phones (for this
)
exten = 100,103,Hangup()
For each extension...
Altus Snyman wrote:
What asterisk version
I know we had a problem with one of the cvs
We couldn't use the transfer buttons,but # worked
What about the Dail(SIP/111,12,tT) in your extensions.conf
On Tue, 2005-02-08 at 13:50, Mark Benson wrote:
I am
release...
Altus Snyman wrote:
What asterisk version
I know we had a problem with one of the cvs
We couldn't use the transfer buttons,but # worked
What about the Dail(SIP/111,12,tT) in your extensions.conf
On Tue, 2005-02-08 at 13:50, Mark Benson wrote:
I am having problems transferring calls from
I have just setup music on hold by downloading and installing mpg123 r
Now I have music on hold but it sounds terrible - clipping, buzzing,
digital distortion, and its too loud (which probably isn't helping) and
I'm just running it thru the 'default' line in music onhold.conf line
default =
Voip to voip (in whatever form that takes sip-sip sip-iax iax-iax) is
what I am using asterisk for.
I would have thought mandrake would have been ok - but haven't used it
for a while. I'm running FC2 (fedora core2) and asterisk complies and
runs without any problems.
Dont fear make. Apps, for
Yesterday I setup music on hold by downloading and installing mpg123 r
Now I have music on hold but it sounds terrible - clipping, buzzing,
digital distortion, and its too loud (which probably isn't helping) and
I'm just running it thru the 'default' line in music onhold.conf line
default =
settings but found nothing odd - anyhoo all sorted now.
Cheers,
Mark
Scott Herrick wrote:
Mark,
I have heard this problem. I'm not exactly sure what the cause is but
check for any duplex mismatches between the phone and the * box.
Hope this helps.
Scott H
Mark Benson wrote:
Yesterday I setup music
CVS in a production environment? Is that advisable?
[EMAIL PROTECTED] wrote:
I am running 1.0.5 and can happily blind xfer from extension to
extension, but I can't blind xfer. I have read various snippets about #2
or #8 or other such key combos, but nothing seems to let me do attended
xfer.
Yesterday I asked about a user manual - ie a user guide to actually
using asterisk (now on how to set it up) the doc project (v2) isn't
anywhere near complete and is the closest thing I could find.
Does anyone know of such a doc? The reason I ask is that while a lot of
this may be obvious to
asterisk -r attempts to connect to a running asterisk process (rather
than starting another one)
-v means be verbose (the more v's the more verbose)
-c provide a control console for asterisk
-g remove resource limit on core size - a debugging thing maybe?
To find all this out for yourself and
I've noticed that too (its not just when having phone sex either! :-).
It depends on the phone being used (or is that abused)?
I have a budgetone that really picks it up and a generic IN1800 (or
something like that) that doesn't pick it up much at all.
And when using soft phones, it depends on
Can you not just remove the sym link to the mpg123 process so asterisk
doesn't find it therefore no music on hold?
When I was trying to get music on-hold working I had to compile and sym
link the mp123 executable - when it wasn't present I had no music on hold...
Mark
MF Hulber wrote:
I'm
I get the impression that the transfer/flash/recall etc etc buttons
don't always work - it seems to depend on what phone/firmware you are
using. And possibly the version of asterisk.
I am using BT102s and some generic voip phone. On the BT102 the transfer
button will put the call on hold and
I'm only just getting my head round asterisk - so the phones themselves
have taken a back seat - I have only recently upgraded the phones to r
.16 - so maybe they do work now. I'll test as soon as all my users have
moved their phones over to the asterisk server.
I only found out about the r
Grandstream are supposed to be releasing a BT103 ? Its a 100 series
phone with headphone jack... when, I couldn't say though.
Thibault Lamy wrote:
Hi there,
Do anyone have any experience with SIP phone that support
a headset ? We have Budgetone phones but we need headsets.
What would you advise
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone lines).
I tried this (from voip-info.org wiki)...
I should mention that I have tried using the call forward function of
the sip phones, but a) this means configuring the phones and some are
remote and behind firewalls and b) It doesn't work...
Cheers,
Mark
___
Asterisk-Users mailing list
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using
Er... set the trunk variable to what? I thought it was a built in
variable...
Peter Bowyer wrote:
Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.
Peter
___
Asterisk-Users mailing list
I have been able to get it working by explicitly setting the dial command...
So should the trunk variable be the divice to dial out on?
Mark Benson wrote:
Er... set the trunk variable to what? I thought it was a built in
variable...
Peter Bowyer wrote:
Have you set the TRUNK variable
Thanks,
Staring to see where I was going wrong. Now I know the explicit dial
string (as you say I tried that in the dial plan and it worked) I can
mess around with the trunk variable.
Cheers!
Peter Bowyer wrote:
On 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
Er... set the trunk variable
http://www.nch.com.au/wavepad/
Brett, Gary wrote:
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them, just
playback the ones that are currently there.
Any help would be greatly appreciated
cheers
How about using a bluetooth headset? You would just need a bluetooth
dongle for the laptop to provide the wireless connection for the headset...
Mark
(i'm in the process of trying this with an old usb bluetooth dongle
(trying to find a suitable driver and manufacturers appears to have
the machine should revert back to the
default hardware for audio playback.
Mark Benson wrote:
How about using a bluetooth headset? You would just need a bluetooth
dongle for the laptop to provide the wireless connection for the
headset...
Mark
(i'm in the process of trying this with an old usb
the button on the headset as well to connect and hey
presto - iTunes on my headset! This works perfectly with xlite to make
and receive calls.
Hope this helps!
Mark
Mark Benson wrote:
If you plan to go this route don't buy a bluetooth adaptor that uses
the XTNDconntect software. I've never been
I wouldn't normally post this to the asterisk mailing list but I'm
really stuck...
I've been trying to get meetme working on and off for a few months now
but I always hit a brick wall when trying to compile.
I keep seeing this...
make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE
Er.. sorry should have thought about this one a bit more - so bogged
down in what isn't working that I forgot that I'm actually trying to
complie ztdummy on a 2.6 kernel...
Mark Benson wrote:
I wouldn't normally post this to the asterisk mailing list but I'm
really stuck...
I've been
Are you using inband DTMF? There are other options but I don't know much
about the polycom phones. I have noticed that sometimes when accessing
voicemail, it will 'miss' some dtmf tones if they are too short. This
doesn't explain the number changing, unless your dial plan is putting in
the
Perhaps he has a server that does other things besides asterisk and
can't reformat it?
Or perhaps he has a server in a remote location and buiness constraints
make it difficult to take the time to get to it and spend a whole day
doing a reinstall?
Mark
Dean Collins wrote:
Dear Sir,
Lol,
Hi all,
I've been trying to get meetme working for a while now (complie problems
- will probably try again later on another machine) but have given up
and started looking at alternatives.
I've managed to get app_conference compiled and installed - show modules
shows its there in asterisk,
exten = 901,1,Conference(Internal Test Conference/S/1)
Looks like it does the job...
Mark Benson wrote:
Hi all,
I've been trying to get meetme working for a while now (complie
problems - will probably try again later on another machine) but have
given up and started looking at alternatives
voiptalk.org seem to be pretty reliable for both incoming and outgoing
calls... I've been using them for at least 6 months for light volume calls.
Steve Foy wrote:
Hi,
I'm looking for a reliable provider to use mainly for outgoing calls in the
UK, incoming isn't so much of a worry as I
Hi I've got an interesting problem.
A few days ago (maybe even a week or two) all my sip phones lost
registrations with my asterisk box. All that is but one.
The asterisk box is out on the internet, I have two phones at my
location and 1 at another separate location.
The only phone that
Hi,
I'm still strugling with getting an easy to use conference system
implemented. I did have app_conference running, but today I upgraded
asterisk to 1.0.9 and it stopped working. I've tried following the
instructions for compiling app_conference on 1.0.7 but it didn't work.
So I went back
and a basic kernel build config file generated.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson
Sent: Friday, September 23, 2005 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ztdummy compile again
Hi,
I'm still
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson
Sent: Friday, September 23, 2005 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ztdummy compile again
When you say kernel development do you mean kernel sources
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