On Sat, Mar 24, 2007 at 12:13:25PM -0700, Martin Joseph wrote:
On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL PROTECTED] said:
Now I know where they've been spending my remaining balance...
I still use Sellvoip as my primary terminator, and have found the call
quality to be superior
So the clocks went forward and I've never bothered to work out the
settings in my grandstream phones - until now!
If you have a Grandstream device it's got the wrong time on it this
morning, then you need to go into the web interface and set
Timezone:
GMT (London, etc.)
Daylight
On Sun, Mar 25, 2007 at 07:43:19AM +0100, Alan Chandler wrote:
Although I have a Debian system with prebuilt asterisk package
available, I was finding it crashed when I tried to use MeetMe.
So I have built asterisk from scratch. However the first thing I try
and do is install the ztdummy
We need incoming calls to simultaneously ring SIP phones, and a cell phone
which is called via a SIP or IAX trunk. When the cell phone answers we'd
like a brief prompt played (e.g. press # to accept call) and if # is pressed
connect the incoming call to the cell phone.
ZAP trunks have some of
Hiya,
i've just noticed that chan_misdn writes the AOCD information into a
logfile. Has someone done a patch that sends this information via sendtext()
to the active channel? At least some phones (like Cisco with chan_sccp and
the snom-phones with SIP) can show this information on the
Hiya,
i've just noticed that chan_misdn writes the AOCD information into a
logfile. Has someone done a patch that sends this information via sendtext()
to the active channel? At least some phones (like Cisco with chan_sccp and
the snom-phones with SIP) can show this information on the
Hiya,
i've just noticed that chan_misdn writes the AOCD information into a
logfile. Has someone done a patch that sends this information via sendtext()
to the active channel? At least some phones (like Cisco with chan_sccp and
the snom-phones with SIP) can show this information on the
A. Levy wrote:
Hello.
I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to
find out if there is any limitation about DSP capabilities, I mean, I
am not sure how many phone calls my Digium card supports,
simultaneously. The calling flow goes from IAX - ISDN.
I am running this
Doug Lytle wrote:
Steve Totaro wrote:
You will probably want some sort or script to reboot the phone
regularly (everyday) or it will just stop working (lose registration
with *). The speaker phones really
Really? I have several of them in use and have yet to reboot any of
them.
Doug
chan_iax2.c does not support *8 pickup groups.
Alvaro Parres wrote:
I had set it
On 3/21/07, LKS GMAIL [EMAIL PROTECTED] wrote:
Try to set the callgroup and pickupgroup up in the IAX conf.
Saludos, Lukassky.
--
*De:* [EMAIL PROTECTED] [mailto:
[EMAIL
Steve Totaro wrote:
I do not deploy toy phones anymore. The % of DOAs and all of the
issues that the BT101s had plus how flimsy it is turned me off. It is
not much more for a Polycom 301 which I consider a true business class
phone. Set it and forget it.
Same here, used them to learn.
I believe they only release the rpm's. It's not open source.
- Original Message -
From: Giorgio Incantalupo [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 23, 2007 6:46 PM
Subject: [asterisk-users] HUD Lite server on Debian
Hi,
anybody knows where to find
On 22 Mar 2007, at 08:30, Tomislav Parcina wrote:
Tim Panton wrote:
I once spent a week struggling with this sort of symptom to
find in the end that the ops guys had got fed up with my
line being in 'alarm' on their console and disabled it at their end.
One phone call later it was re-enabled
I have an interesting task for my son's lacrosse team... it is the
time-old telephone tree...
I am pretty sure someone has already done this w/*, why re-invent the
wheel?...
a) coach calls in leaves a msg, others call in retrieve the msg
b) coach calls in leaves a msg, kicks of a call to
Gordon:
I was all excited to see a reply to my message with the subject
TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC -- but
instead I got this:
Gordon Henderson wrote:
So the clocks went forward and I've never bothered to work out the
settings in my grandstream phones - until
Just a partial answer:
On Sat, Mar 24, 2007 at 10:24:18AM -0600, Stephen Bosch wrote:
Hi, everyone:
I am developing a system using Asterisk, TDM-400 analog cards, analog
lines, and Polycom SIP phones for internal extensions.
Initially there was bad echo but after a series of efforts, I've
My experience has been decent with vitelity.net. Not stellar, but fairly
responsive customer support as long as I use their support ticket
system, I'd usually have a response within one hour during business
hours. They did get a little testy with me when I complained about a
port taking nearly 60
Check Nerd Vittles. They even include it in their installation scripts.
Thanks,
Steve
dave cantera wrote:
I have an interesting task for my son's lacrosse team... it is the
time-old telephone tree...
I am pretty sure someone has already done this w/*, why re-invent the
wheel?...
a) coach
I just upgraded to asterisk-1.2.14 and using default streamplayer
though, I don't think is has anything to do with the voice messaging
system, does it?
When I enter the mailbox to listen to the recored message I press 1
and when the message starts playing all it plays is:
First messge received
On Sunday 25 March 2007 11:07, Tzafrir Cohen wrote:
On Sun, Mar 25, 2007 at 07:43:19AM +0100, Alan Chandler wrote:
...
aptitude search zaptel | grep ^i
Ah - I had no idea I needed this zaptel package. the Asterisk package
doesn't even have it as a suggests!.
Actually, this looks more
Nothing has changed in my Asterisk configuration and now outbound US is
getting nothing, but 403's. Anyone else having the same problem? Inbound
calls to my DID's are working fine.
Thanks, SG
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Joseph wrote:
The error message I get:
Mar 25 11:39:02 WARNING[16945]: file.c:824 ast_streamfile: Unable to open digits/1 (format ulaw): Permission denied
Mar 25 11:39:08 WARNING[16945]: file.c:824 ast_streamfile: Unable to open
digits/at (format ulaw): Permission denied
It's a
Stephen Bosch wrote:
I'm not questioning the value of your contribution, I'm just suggesting
it should be in its own thread :)
HERE HERE!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
On Sun, Mar 25, 2007 at 07:14:22PM +0100, Alan Chandler wrote:
On Sunday 25 March 2007 11:07, Tzafrir Cohen wrote:
On Sun, Mar 25, 2007 at 07:43:19AM +0100, Alan Chandler wrote:
...
aptitude search zaptel | grep ^i
Ah - I had no idea I needed this zaptel package. the Asterisk package
Hi:
Tzafrir Cohen wrote:
Just a partial answer:
On Sat, Mar 24, 2007 at 10:24:18AM -0600, Stephen Bosch wrote:
Hi, everyone:
snip
Now, standard analog sets have a varistor circuit to compensate for
these variations in signal level, but it would appear that the TDM cards
don't incorporate
Salvatore Giudice wrote:
Nothing has changed in my Asterisk configuration and now outbound US is
getting nothing, but 403’s. Anyone else having the same problem? Inbound
calls to my DID’s are working fine.
Clearly, sellvoip rocks!
-stephen-
___
Hi dave,
Sorry I missed your mail. My system is a production system. Any advice ?
Thank you
- Original Message -
From: dave cantera
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, March 21, 2007 11:37 AM
Subject: Re: [asterisk-users] Fedora +
I have implemented the requested call confirmaiton feature in the freepbx
followme and ringgroup applications (asterisk 1.2 for now). You can select to
have confirmation and by default any external call (e.g. cellphone) will
require such confirmation, any internal phone will not (unless you
marek cervenka wrote:
please look at
http://www.voip-info.org/wiki/view/Asterisk+SRTP
and try compilerun clients with srtp (linksys,gxp-2000, minisip,
twikle, )
Does it work between two asterisk? If I use
gxp-2000 = * = * = phone that doesn't support SRTP
will it work?
--
Tomislav
On Sunday 25 March 2007 19:33, Tzafrir Cohen wrote:
On Sun, Mar 25, 2007 at 07:14:22PM +0100, Alan Chandler wrote:
On Sunday 25 March 2007 11:07, Tzafrir Cohen wrote:
On Sun, Mar 25, 2007 at 07:43:19AM +0100, Alan Chandler wrote:
...
aptitude search zaptel | grep ^i
Ah - I had
We regularly install * on Fedora (clients with lots of leading edge hardware
like Fedora). No problems
I expect you will only encounter * 1.4.x errors like everyone else.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit
The issue with FreePBX is that it uses the Asterisk database to store user and
device information (e.g. who is the currently logged-in user). So you need to
replicate that information across multiple machines.
The approach we have taken is to customise FreePBX (not trivial) so that all
this
On Sun, 2007-03-25 at 14:29 -0400, Doug Lytle wrote:
Joseph wrote:
The error message I get:
Mar 25 11:39:02 WARNING[16945]: file.c:824 ast_streamfile: Unable to open
digits/1 (format ulaw): Permission denied
Mar 25 11:39:08 WARNING[16945]: file.c:824 ast_streamfile: Unable to open
Steve Totaro wrote:
Jay Milk wrote:
Doug Lytle wrote:
Jay Milk wrote:
I've done all the googling I can on this, and have come to the
conclusion that a Grandstream BT101 can be abused to be a door
phone. Could someone with access to one, confirm that the
following is possible?
Researched:
marcotasto wrote:
I did something similar one year ago for a friend of mine that was interested
to answer to bell door from internal phones.
I used an HT286 with a sort of homebuilt analog hybrid with a microcontroller
able to automatically answer when the ring was present on the HT286 FXS
Jay Milk wrote:
Steve Totaro wrote:
Jay Milk wrote:
Doug Lytle wrote:
Jay Milk wrote:
I've done all the googling I can on this, and have come to the
conclusion that a Grandstream BT101 can be abused to be a door
phone. Could someone with access to one, confirm that the
following is
Steve Totaro wrote:
Just get a Grandstream ATA and a handset with no buttons. So simple.
Do you know of any vendors with inexpensive handsets without buttons?
___
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asterisk-users mailing list
To
I'm not surprised.
On 3/25/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Salvatore Giudice wrote:
Nothing has changed in my Asterisk configuration and now outbound US is
getting nothing, but 403's. Anyone else having the same problem? Inbound
calls to my DID's are working fine.
Clearly,
Eric ManxPower Wieling wrote:
Steve Totaro wrote:
Just get a Grandstream ATA and a handset with no buttons. So simple.
Do you know of any vendors with inexpensive handsets without buttons?
I know of some that have the buttons on the cradle and the hook switch
on the handset, that would
HiI don't have any digium cards and only want to install ztdummy with all
asterisk functions. What I will need with ztdummy and what I can disable?Thanks
a lotDmitri
Zaptel Module Selection
Eric ManxPower Wieling wrote:
Steve Totaro wrote:
Just get a Grandstream ATA and a handset with no buttons. So simple.
Do you know of any vendors with inexpensive handsets without buttons?
I know of some that have the buttons on the cradle and the hook switch
on the handset, that would
Hi All,
Can anybody guide me to check the D-channel info of my netjet ISDN card
I am trying to configure it with asterisk using misdn_capi and chan_capi.
How can set differnt protocol at D-channel.
Thanks Regards
Farooq
--
___
--Bandwidth and
I ran into a problem today while trying to compile chan_cellphone version 17
on a CentOS 4.4 machine. Apparently the bluez and autoconf versions were to
old and as I tried to install the latest version, I found that the new
bluez-lib would install and allow the chan_cellphone to compile, but
On Sun, Mar 25, 2007 at 10:18:03PM +0100, Alan Chandler wrote:
On Sunday 25 March 2007 19:33, Tzafrir Cohen wrote:
On Sun, Mar 25, 2007 at 07:14:22PM +0100, Alan Chandler wrote:
On Sunday 25 March 2007 11:07, Tzafrir Cohen wrote:
On Sun, Mar 25, 2007 at 07:43:19AM +0100, Alan Chandler
On Sun, Mar 25, 2007 at 09:47:28PM -0400, Dmitri Smirnoff wrote:
Hi
I don't have any digium cards and only want to install ztdummy with
all asterisk functions. What I will need with ztdummy and what I can
disable?
You only need zaptel and ztdummy of the modules . You don't need any of
the
On Sun, Mar 25, 2007 at 04:03:42PM -0600, Joseph wrote:
On Sun, 2007-03-25 at 14:29 -0400, Doug Lytle wrote:
Joseph wrote:
The error message I get:
Mar 25 11:39:02 WARNING[16945]: file.c:824 ast_streamfile: Unable to open
digits/1 (format ulaw): Permission denied
Mar 25 11:39:08
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