Hello,
Does Asterisk-GUI 2.0 compatible with Asterisk 1.6.1.1 ?
Sincerely,
Tseveen
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Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
I presume the same rules apply for scaling and possibly have
OpenSIPS/Kamailio on the front?
Thanks.
--
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On Fri, 31 Jul 2009, Gavin Henry wrote:
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
Intersting. Looks like BT trying to become an ITSP to compete with the
other ITSPs in the UK who already have PSTN
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, Gavin Henry wrote:
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
Intersting. Looks like BT trying to become an ITSP to
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, Gavin Henry wrote:
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
Intersting. Looks like BT trying to become an ITSP to
On 31 Jul 2009, at 08:22, Gavin Henry wrote:
Has anyone passed the tests using Asterisk:
BT guy we spoke to said yes : )
Steve
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Hi,
I think there is an essential option of the Voicemail application that
is missing.
I would like to suggest the implementation of a function to give the
user the ability to either allow or disallow the recording of messages.
If the ability to record a message is disabled, options u, s, and b
So made my first forray into 1.4 and DAHDI and hit a problem. (Not
convinced this is a DAHDI issue though...)
Testing an analogue line and asterisk sees the caller ID being passed, but
then fails to detect ringing. A plain old analogue phone plugged in rings
just fine.
Console output:
==
- Gordon Henderson gordon+aster...@drogon.net wrote:
So made my first forray into 1.4 and DAHDI and hit a problem. (Not
convinced this is a DAHDI issue though...)
Testing an analogue line and asterisk sees the caller ID being passed,
but
then fails to detect ringing. A plain old
Hi jonathan
thx for the tips but with your solution the user can only log from one
computer. .
I thinks your solution may be usefull if the user use a physical phone but
if he use a softphone he became dependent of the IP adress. :-(
but I may have another way : get informations from asterisk
2009/7/31 Steve Howes st...@geekinter.net:
On 31 Jul 2009, at 08:22, Gavin Henry wrote:
Has anyone passed the tests using Asterisk:
BT guy we spoke to said yes : )
Good to know!
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http://www.suretectelecom.com
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my extensions.conf
...
; Unconditional Call Forward
exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten = _#21*X.,2,Hangup()
exten =
Hi,
I was testing failover trunk with one IP trunk and 1 E1 trunk
IP trunk is the primary trunk and e1 is secondary.
I block connection to test failover for this system.
I got the msg unreachable for my IP trunk on the system as warning.
Then i tried to dial to an extension outside.
It waited
On Fri, 31 Jul 2009, --[ UxBoD ]-- wrote:
Gordon,
Cast your mind back as I had a similar issue ... changing the cable sorted it
for me!
Cursiously enough, I thought about that - but these were 2 brand new
cables out of packets and I did check to see that they only had 2 wires
connected at
Emrah wrote:
Hi,
I think there is an essential option of the Voicemail application that
is missing.
I would like to suggest the implementation of a function to give the
user the ability to either allow or disallow the recording of messages.
If the ability to record a message is disabled,
Hi,
Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a
lot of experience in the telecommunications space and he joins us
today to chat about its current state, conferencing and whatever else
comes to mind. So we have a meta conference aout conferencing, it
won't be the first
It does seem like a simple thing to accomplish. Since Asterisk is
open-source, you should give it a whirl or post a bounty. You could
accomplish the same thing in the dialplan, but that wouldn't be a feature.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a
lot of experience in the telecommunications space and he joins us
today to chat about its current state, conferencing and whatever else
comes to mind. So we have a meta conference aout conferencing, it
won't be the first
Anyone have a customer sending/receiving multi-page faxes over Verizon
Business SIP trunk/g711 ?
Verizon Business indicates they don't support it, and I have 2 recent
customers that it doesn't work for, and 1 current large customer telling
me he's going to make it work grin.
The issues is
Andrew Thomas wrote:
[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
deny=0.0.0.0/0.0.0.0
allow=xxx.xxx.xxx.0/255.255.255.0 read what you've put!!! The
'allow' should be 'permit' as Jared already told you (and he should know
what he's talking about).
insecure=port,invite
Mark,
I think you did not understand my message.
I am accustomed to have the option to allow or disallow the recording of
a message in my voicemail, even my mobile carrier provides it. E.g.: I
can record a message that says Please call back later, I am currently
on the phone. without any beep
How 'bout setting up an extension that simply plays an announcement and
hangs up. Then transfer calls from extensions that don't want messages to
this extension.
You could have a few extensions with a few different recordings to suit
different situations.
--Don
Don Kelly
PCF Corp
People Come
Emrah wrote:
Mark,
I think you did not understand my message.
I am accustomed to have the option to allow or disallow the recording of
a message in my voicemail, even my mobile carrier provides it. E.g.: I
The simplest thing to do is to allow users to set a flag, maybe using
mysql or
Solved..
just add qualify=yes to the trunk config.
Hi,
I was testing failover trunk with one IP trunk and 1 E1 trunk
IP trunk is the primary trunk and e1 is secondary.
I block connection to test failover for this system.
I got the msg unreachable for my IP trunk on the system as warning.
I have problems with it...
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler:
Found license 'XX' providing 1 concurrent calls
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype
For Asterisk Host-ID: X
[Jul 30 14:34:17] NOTICE[30529]:
Wow, been a long time since I have been on the list.. A few years to be
exact :)
Glad to be back in the land of Asterisk..
I have a box running Asterisk 1.4.8 that's been real solid and I have a
bunch of custom stuff running on it.
I am trying to move this to a new piece of hardware and
Are your permissions ok (files and directories)? Did you restart asterisk
after modifying moh.conf?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier
Sent: Friday, July 31, 2009 1:49 PM
To:
Looking at the code, asterisk doesn't know how to handle RPID in an INFO
message, so it just responds with an OK and goes on with it's business...
The fact that the message has the name of the called party, rather than the
calling party probably wouldn't help even if Asterisk did understand it...
It was the darn restart, thanks!
I wasn't aware that musiconhold.conf was only read at start but now I
remember that Asterisk only reads certain files upon reload.
Thx
On Jul 31, 2009, at 2:03 PM, Danny Nicholas da...@debsinc.com wrote:
Are your permissions ok (files and directories)? Did
2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
On Fri, 31 Jul 2009, --[ UxBoD ]-- wrote:
Gordon,
Cast your mind back as I had a similar issue ... changing the cable sorted
it for me!
Cursiously enough, I thought about that - but these were 2 brand new
cables out of packets and I
Doug,
Thanks for the suggestion.
I know there are plenty of workarounds there, I am not asking how to do
it because I know how to do it too.
What I am saying is that it could be an embedded feature in the
Voicemail application, like the recent ability to flag a message as
urgent.
Regards,
Emrah
Dom,
Quoting myself from your original message:
I know everything is possible to be done via the Dialplan. I could just
have a Playback to achieve this or pretty easily code my own voicemail
app via AGI too.
I am not asking how to do this. I know everything is possible with Asterisk.
In my
Gavin Henry wrote:
Hi All,
Has anyone passed the tests using Asterisk:
http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html
I presume the same rules apply for scaling and possibly have
OpenSIPS/Kamailio on the front?
Thanks.
We have, asterisk 100% +
Dear David
I appreciate your reply . Pleae find attached our current extensions.conf
file . Can you please do me favor and let me know where I am expected to put
your proposed line for defining the timeout param ?
Regards
H.Motamedi
On Fri, Jul 31, 2009 at 3:16 AM, David Backeberg
Dear All
Please be informed that we have an application for our subs to be able to
dial #21 to reach IN services . Can you please let us know how we can
support for this as it seems that the Asterisk does not support for the hash
# key as an valid extension to be dialed by the user ?
Regards
Dear All
Can you please let us know how to configure Asterisk to recognize extensions
starting with the hash key ?
Regards
H.Motamedi
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