First of all test your odbc-connection via console:
isql telco-ops dba c3podb@2012 -v
You should see a Connected!-Message. Do you?
Second: yes I also had problems setting up odbc. The main
problem/error for me was, that documentation is sometimes confusing.
Here is my config. Please notice the
On Tuesday 11 December 2012, Joseph wrote:
On 12/10/12 20:45, Steve Edwards wrote:
On Mon, 10 Dec 2012, Joseph wrote:
When a call comes in asterisk records the date correctly but when I cake
a call out I get only something like:
Date: 60
here is an example:
From: 7807560785
On Mon, Dec 10, 2012 at 11:02 PM, Joseph syscon...@gmail.com wrote:
On 12/10/12 20:45, Steve Edwards wrote:
On Mon, 10 Dec 2012, Joseph wrote:
When a call comes in asterisk records the date correctly but when I cake
a
call out I get only something like:
Date: 60
here is an example:
On Tuesday 11 December 2012, Dale Noll wrote:
. [stuff deleted] . You could
also write a script around what you are doing and parts records differently
based on the 'last appliation' such as Dial, Read, Voicemail.
Prioblem: You still don't know how many commas are going to be
On Tue, Dec 11, 2012 at 10:07:42AM +0530, Harish Mandowara wrote:
Hi,
Thank you for your reply.
77 ext. number is connected with my asterisk. so any one want to
talk with jitsi(pc), they have to dial 77 then 2000#(jitsi sip
user number).
my pbx is sending callerid. i can see on other
Dear List,
Where can I find a guide for setup an Asterisk server which can eastanblish
a simple video call from two sip clients?
Thank you!
Regards,
Barco
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Most of the time my phone line are working OK but at time to time when I run:
asterisk -rx core show channels it show:
Channel Location State Application(Data)
SIP/pstn--00 (None) Up AppDial((Outgoing
Line)) SIP/pstn-9998-00
Could be, but I'd check the easier to fix polarity settings.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Tuesday, December 11, 2012 11:27 AM
To: asterisk-users@lists.digium.com
Subject:
On 12/11/12 11:30, Danny Nicholas wrote:
Could be, but I'd check the easier to fix polarity settings.
How do I do that?
Notice, that this channel hang-up/disconnect does not happen all the time, only once a while could be once a day or once a week.
--
Joseph
--
In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and
hanguponpolarityswitch lines. If they aren't present, the default values
are being used. If they are, tweak them and restart asterisk and dahdi. I
do this - service asterisk stop; service dahdi restart; service asterisk
start.
On 12/11/12 11:48, Danny Nicholas wrote:
In /etc/asterisk/dahdi.conf, check your answeronpolarityswitch and
hanguponpolarityswitch lines. If they aren't present, the default values
are being used. If they are, tweak them and restart asterisk and dahdi. I
do this - service asterisk stop;
You need to look at the device which the analog lines plug into. There is
nothing to change in Asterisk for this issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Tuesday, December 11, 2012
If you are using an analogue/sip ata, then the problem is on the ata. Run a
packet capture and you'll see the invite coming from the ata without nobody
using the phone...
I am typing from my mobile phone...
Il giorno 11/dic/2012 18:55, Joseph syscon...@gmail.com ha scritto:
On 12/11/12 11:48,
I have an asterisk server at home. I'm looking to replace my internal
phones with sip cordless (DECT) phones. I'm now looking at the Siemens
A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base
($80) and DP710 handset ($45).
The Siemens has a feature were I can also use a
I've been using the Gigaset A580 Base and A58H Phone for about 3 years
now. Never gave me problems. The call Quality is excellent!
I only have 1 handset connected to the Base but I want more. I bought a
Linksys WIP330 as a 2nd phone to try out and that works just as good
without a base unit.
On Tue, 2012-12-11 at 23:02 +0800, Barco You wrote:
Dear List,
Where can I find a guide for setup an Asterisk server which can
eastanblish a simple video call from two sip clients?
Thank you!
Regards,
Barco
Hi Barco,
I don't think there is a specific guide for this.
From the
One thing I dislike about the A580H is that the handset always says 'You have
new messages' if I've missed a call. It wouldn't bug me if it said 'missed
call' but it tells me I have new messages and even lights up a red LED under a
button with a picture of an envelope on it.
I'm about to test
That is true about the A580.
I don't like the interface much to check messages.
Besides that every time I go to dial a number...it always uses the first
digit pressed to go into phone mode..so I have to press the first digit
twice...
I would test other phones but it's for home and I can't
Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking for a way
to determine the name of the calling context.
--
Mitch
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New to Asterisk? Join
You don't state version, but I'm pretty sure this animal doesn't exist.
What I did in 1.4 was to set a variable before the gosub so I could track
it. Something like this
Exten = s,n,Set(from=foo)
Exten = s,n,gosub(showfoo,s,1)
Exten = s,n,Set(from=bar)
Exten = s,n,gosub(showfoo,s,1)
[showfoo]
On 12/11/2012 04:37 PM, Roy Abshire wrote:
That is true about the A580.
I don't like the interface much to check messages.
Besides that every time I go to dial a number...it always uses the first
digit pressed to go into phone mode..so I have to press the first digit
twice...
I would test
Was looking for 1.8 and above. I ended up doing something similar to
what you describe. Not terribly elegant, but it works.
Mitch
On 12/11/2012 04:03 PM, Danny Nicholas wrote:
You don't state version, but I'm pretty sure this animal doesn't exist.
What I did in 1.4 was to set a variable
I never hear a dial tone from my sip phones. I just pick up the phone
and dial + send but you should be able to dial out using the same SIP
account in use...but you will need at least 2 outgoing trunks with your
SIP provider to call external numbers, unless your calling another
extension.
Using a Gigaset C610IP here, and am very happy with the features. The base
station can handle two concurrent SIP calls, and another internal one at
that. It does it with a single SIP registration to each server. You can
setup multiple servers if you want to and define dial patterns/plans that
Mebbe you guys should try snom m9 dect ip phone, i have been using it since
over 3 years now without any of these issues.
Mitul
On Dec 12, 2012 4:25 AM, Kai-Uwe Jensen kujen...@gmail.com wrote:
Using a Gigaset C610IP here, and am very happy with the features. The base
station can handle two
At the risk of continuing off-topic conversation...
Oh the M9 has it's own issues, don't you worry (not to mention it's _way_ more
expensive than the Gigaset range)!
I've been testing the A510 today and I've decided I like it more than the A580.
The software (via the web interface) looks more
Pete,
Thanks for testing these out. You said you like the 510 vs 580. Which
one is newer? I have the 580. I'm staying tuned for your review of the
610. The 580 isn't DECT compatible and only supports some of the
Gigaset handsets.
I might sell mine and upgrade depending on what you learn
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