Hi,
We are getting a pulsing clicking sound when using
siptone phones on a customer network. The called person hears pulsed clicking
sounds, whereas the person on the siptone phone hears pulsed hissing. We are
using alaw all the way through and there is no codec conversion taking
place.
Hi,
We had a
problem this evening where asterisk was running but unable to receive any iax or
sip traffic. We restarted * and then it was fine. In hindsight we should have
assessed whether this was a deadlock situation.Has anyone else seen this
problem? We're using the -head cvs of April 2
Hi Frederic,
> What does not work though is when the phone is ringing, nobody else can
call the phone anymore.
By this, I presume you mean when the phone hasn't yet been answered?
We are seeing problems where if a snom user is on the phone and another call
comes in, then the person he/she was spe
There is no need to use oh323. If you look in
/usr/src/asterisk/channels/h323 then you will find that there is already an
h323 implemenatation present (chan_h323). You just need to follow the
instructions and it works great.
Tan
telappliant.com
-Original Message-
From: [EMAIL PROTECTED]
Hi,
Unless there has been a recent change, you can't set codecs in the sip.conf
on a per-context basis. The way to do what you want is to have the following
in the [general] area:
disallow=all
allow=ulaw
allow=alaw
allow=g729
Then, set all the codec preferences on the g729 phone to g729. That ph
Only
blind transfer I think. Attended transfer can be sort of be done using asterisk
call parking feature.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Bartosz
JozwiakSent: 01 October 2003 20:04To: ASTERISK
USERSSubject: [Asterisk-Users
Check
http://www.telappliant.com. Quantity
discounts available. Shipping from the uk.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Michael
KoehlerSent: 16 September 2003 09:43To:
[EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Re:
We have successfully tested isdn bri with DDIs, CID for one of our
customers. Basically they have 10 uk DDIs routed to a standard isdn 2 line.
Each ddi looks like a separate extension to *, and we also have a fax
capability which was configured using hylafax. Works well.
Tan
TelAppliant Ltd
Are you sure that your telco hasn't busied one of them out? We had a similar
problem.
- Original Message -
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]>
Sent: Wednesday, June 04, 2003 10:46 AM
Subject: [Asterisk-Us
rst few seconds of
every call being broadcast.
Am I missing a setting on the 100 to make the headset selection more
permanent? Does the 200 support headsets after all? Properly?
Many thanks,
Simon
- Original Message -
From: "T Aksoy" <[EMAIL PROTECTED]>
To: <[EMAIL
Our prices (for singles):
SNOM 100: £169+VAT (free delivery)
SNOM 200: £189+VAT (free deliver)
Personally (as I'm sure the guys on this group would agree) I would go for
the SNOM 200. Look on the emailing list for a lengthy discussion on this
subject.
Tan
- Original Message -
From: "n
Hi Michiel,
How much do you think it would be? Since we are planning on mass sales in
Europe then I may be interested.
Thanks
Tan
- Original Message -
From: "Michiel Betel" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, April 05, 2003 11:19 PM
Subject: RE: [Asterisk-Users
Cygwin does have the cvs command, but it isn't installed as default. Go back
and tick more option when doing your install.
Tan
- Original Message -
From: "it" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, April 02, 2003 5:41 PM
Subject: Re: [Asterisk-Users] How could I get
o.
>
> Andre
>
> - Original Message -
> From: "T Aksoy" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, March 30, 2003 12:23 PM
> Subject: Re: [Asterisk-Users] SNOM200 Message Waiting works with current
CVS
>
>
> > Hi,
&g
Hi,
Yes, I can confirm that too. But when the I press the button marked "MWI",
it just says "Not Found". It looks like it is trying to dial
[EMAIL PROTECTED] instead of my voicemail extension which is 8500.
Where do I change this?
Thanks
Tan
- Original Message -
From: "Andre Bierwirth"
I prefer the 200, mainly because of its more comforable handset. It feels
more like a conventional telephone.
Most of the 200's configuration is done via the web interface, whereas the
100 has a lot more control via the lcd display. Also, the 200 supports an
external headset. If you want to test h
How would I go about implementing a voicemail callback
feature. The idea is that asterisk contacts any extensions that have voicemails
waiting, and plays them back. There would have to be parameters such as number
of callback retries, time between retries.
Thanks
Tan
This works for me on the snoms and I get callerid. But when I call from a
snom to our trial sip phones, all that comes on the screen is "asterisk
calling". Any ideas as to where the word "asterisk" is being picked up from?
T
- Original Message -
From: "Andre Bierwirth" <[EMAIL PROTECTED]
The thing that we have noticed is that there is quite high voice
attenuation. Adjusting the txgain and rxgain just seems to cause distortion.
Tan
- Original Message -
From: "Jim Archer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, March 16, 2003 12:31 AM
Subject: RE: [Aster
Hi,
We are trialling a new sip phone and I am having
difficulty getting the voicemail to recognise the dtmf tones. I have tried
setting the dtmf mode on the phone to out-of-band but still no
luck.
How can I determine which end (asterisk or phone)
has the problem?
ThanksTanTelappliant
S
Hi,
We are testing a number of sip phones from
different manufacturers. With one phone in particular, when I dial the asterisk
voicemail, it misses around half a second from the beginning of the
announcement. I don't have this problem with the snom 200 or 100.
Does anyone know why this ha
rxgain and txgain are in db.
We have a similar problem which is even more noticeable since we divert
calls by receiving on one fxo card #1 and sending out on fxo card #2. I
can't seem to find a properly working solution for the attentuation which is
taking place.
For your issue, try setting txgai
Hi Mark,
Not familiar with "received=". What does it do? Has it got any application
within the nat domain?
Thanks
Tan
- Original Message -
From: "Mark Spencer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 11, 2003 8:25 PM
Subject: [Asterisk-Users] FIX: iconnect + del
sage -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, March 09, 2003 2:02 AM
Subject: Re: [Asterisk-Users] Verbose setting changed?
how about "sip debug" ?
regards
Martin
On Sat, 8 Mar 2003, T Aksoy wrote:
> Hi,
>
> O
Hi,
On the release of asterisk I was using before this
one, I used to issue a "set verbose 100" command and I would see all the sip
registrations taking place. Now that doesn't seem to work.
Could someone clarify what value I should use with
the "set verbose" command in order to see sip r
Finally someone has hit the same problems that we have. Everyone on this
newsgroup seems to have static IPs!
The problems you get can manifest in 2 ways:
1) you cannot get through to the phone at all
2) one-way audio - you can hear the other end but they can't hear you.
The problem is a combina
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