Please stop plugging the book. Its annoying. We know
its out there.
Dovid
--- Leif Madsen [EMAIL PROTECTED]
wrote:
On 1/16/06, John Falk [EMAIL PROTECTED] wrote:
Can someone show me how to set up DUNDi, I will be
using it to connect
14 asterisk servers internally. I don't want to
use it
Hello List,
I have more of a generic question. A lot of times when
links to books, little bits of codes, diffrent
programs etc. are posted I do a wget to my server so I
can have it for future yes. Every now and then I reply
to questions with links to these kinds of things. I
have never posted the
Dont think one exists. You may want to get an ATA that
has 24 FXS ports on it.
Regards,
Dovid
--- Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for a PCI card which i could install on
asterisk box, with
purpose to use 15-20 cordless dect phone in a very
dect cell.
Is
]
[mailto:[EMAIL PROTECTED]
On Behalf Of
Dovid Bender
Sent: Sunday, January 22, 2006 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Gen. Question
Hello List,
I have more of a generic question. A lot of times
when links
to books
As a general rule if the phone is behind NAT there
should be no issues. Server behind NAT = Lots of
issues (which can all be worked out). You will have to
specify NAT=YES in the dial plan.
Regards,
Dovid
--- Moises Silva [EMAIL PROTECTED] wrote:
you can redirect the ports of the router as
I was waiting for the spam to start... Looks like
were off
--- Laurie Whitaker [EMAIL PROTECTED]
wrote:
China: The World's New Engine for Growth
Everywhere smart investors have been accumulating
positions in China
related equities, with the knowledge that this vast
new market is
es iz nisht shain tzoo shrieben azelecha verter
--- C F [EMAIL PROTECTED] wrote:
wtf
On 1/24/06, Laurie Whitaker
[EMAIL PROTECTED] wrote:
China: The World's New Engine for Growth
Everywhere smart investors have been accumulating
positions in China
related equities, with the
Let me guess you have no affiliation with them what so
ever ? no commision on accounts either ?
--- Kaleb L. Kunzler
[EMAIL PROTECTED] wrote:
I use iax.cc and find their service to be superior
to ANY other VOIP
provider I have tried. Their prices are
competitive, My calls always go
through,
When you open your burning software there should be an
option to burn from an image. When it asks you for the
location tof the image point it to the .iso file that
you downloaded. After it is done burning the CD you
have a ready to go bootable CD. BE CAREFULL. Once you
put the CD into a machine it
I know this may be a backwards way but for several
reasons I have asterisk send all calls thru astcc.
With astcc you specify multiple routes with prioroty
settings. If it cant complete a call with one route it
will roll over and use the next one.
Regards,
Dovid
--- Cavanna, Richard [EMAIL
A work around can be that when asterisk restarts it
sends a signal to the SNOM to restart.
Regards,
Dovid
--- [EMAIL PROTECTED] wrote:
On Thu, 2006-01-26 at 15:31 +, c waddy wrote:
We are looking to replace our existing Legacy
PBX with Asterisk. Our
receptionist currently has a
If you get it working please let me know how you do
it. I have tried multiple times with no success.
Regards,
Dovid
--- Mike Hammett [EMAIL PROTECTED] wrote:
I'm running a VPS and I need to pass the device
drivers from the host OS to the VPS. What files do
I need to pass through for ztdummy
I would like to add that I did have at one point
problems figuring out 4.0 and there were no problems
downgrading. Also I made a special email account
@mydomain for SNOM liscence's. This helps if at a
later dat you need to re-enter it again.
Regards,
Dovid
--- Christian Stredicke [EMAIL
To monitor who is doing what we writing a program that
every user can have on thier windows desktop to see
the status of all phones on the system. It's AIM
style. Has several groups. On the phone, off,
Available, Away etc.
Managers can scroll the mouse over the user and see
what call they are on
Generaly you get what you pay for (with very few
exceptions such as asterisk). Also as far as a web
interface goes its really one that you get used to and
like. There are lots out there. You goto find one that
works for you.
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Jan 31, 2006 at
A)When you say stop asterisk from transfering the call
what do you mean ? oNot to send it to VM if the user
is away ?
B)I think it depends on the phone. I know with the
Polycoms you can program it directly in to the phone.
(Done it in the past).
--- Bartosz Piec [EMAIL PROTECTED] wrote:
A user
Not sure if this will help but for multiple reasons we
send all calls thru astcc. In astcc you specify what
route you want it to use. If a route isnt available
then it tries the next one you specified and so on.
--- Jolly M. Recto [EMAIL PROTECTED] wrote:
Hi,
i have diffirent provider
From simulas posts as the one you have written it
seems they are heading in the path of live voip. They
grew too big too fast and cant handle customer
service. I would recomend using myPhoneCompany. One
Negative note, at this time for unlimited plans I know
they only offer ATA's. If you want you
: SHA1
care to share with the rest of the class?
Dovid Bender wrote:
To monitor who is doing what we writing a program
that
every user can have on thier windows desktop to
see
the status of all phones on the system. It's AIM
style. Has several groups. On the phone, off,
Available, Away
From simulas posts as the one you have written it
seems they are heading in the path of live voip. They
grew too big too fast and cant handle customer
service. I would recomend using myPhoneCompany. One
Negative note, at this time for unlimited plans I know
they only offer ATA's. If you want you
iBell just announced termination only to CA for I
believe $0.0039 a minute.
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On Thursday 02 February 2006 07:39, hugolivude
wrote:
I'm looking for a new Internet Telephony Service
Provider for my company in
Canada to terminate calls from my
Anyone know of any equipment that I can use to connect
a laptop running asterisk to a POTS line (RJ11) ?
Regards,
Dovid
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Thursday, February 02, 2006 6:20 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk on laptop
connected to POTS line
Anyone know of any equipment that I can use
Yes. The wiki and voip-info.org
--- Zach A [EMAIL PROTECTED] wrote:
Hi,
Is there any detailed guide/tutorial source online
on queues?
Zach
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To
I believe that this email should be going to the BIZ
list only.
--- Sam Tam [EMAIL PROTECTED] wrote:
Single port GSM Gateway support 900 / 1800 GSM mode
with external antenna.
Brand new unit and all of them will be tested before
dispatch.
Extremely easy to setup and can be used
This is the second time this week. This email belongs
on the BIZ LIST ONLY.
--- Sam Tam [EMAIL PROTECTED] wrote:
The long waited Ultimate GSM Gateway is finally out.
This time we have managed to source a new patch of
brand NEW GSM Gateway at prices that is only 50% of
what the market rate.
Hello list,
I am currently doing a job for a summer camp. They
would like to have several phones around the camp from
which people can call in to the main office. It is an
older campus and it is comprised of mostly old
nungalow type housing. I need to install these phones
several hundred yards
For prepaid billing I would use astcc. It comes with
asterisk (it may be in an extra download but is made
by digium to work with asterisk). For post billing
there are a lot of diffrent solutions out there. Do a
google search. As far as putting in the DID info you
would put it in to your dial plan.
I think your problem is the Dell 650. What are the
specs on it ? If you want a system that can support
200 users you will need to do a lot better than that.
Also you will be dealing with T1's/E1's and not POTS
lines. I think a good place to start (if you havent
already) is the book that has come
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Dovid
Bender
Sent: Thursday, February 09, 2006 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Asterisk vs.
Traditional PBX
I think your problem is the Dell 650
I am actually now working on massproducing door
openers that will work with asterisk. It will have an
rj45 port and then a port to plug the door opener in
to. Please contact me off list if you are interested.
Dovid
--- Thomas Artner [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Maybe do a
Voipjet should be fine however dont have all the calls
go out at once. test your system first and see how
many concurent calls you can have at once without
loosing voice quality. I would also reccomend getting
a dedicated server to do the calls as apposed to
buying the equipment if you are doing a
why not try www.asteriskhelpdesk.com
(No, I am not affiliated with them in any way)
--- Dean Collins [EMAIL PROTECTED] wrote:
Hi all,
I was just on the phone with a B2C company in
Australia who are looking
to integrate an Asterisk solution with their
Salesforce.com CRM
platform.
I have a half system thats almost done. Had a client
that wanted it then backed out. Please contact me off
list.
Dovid
--- Michael Sampson [EMAIL PROTECTED] wrote:
Does anyone have any system in place that does
automated wake up calls.
With recordings and options configurable over the
I may be missing something here but why wouldnt ATA's
work ? (other than cost).
--- maka [EMAIL PROTECTED] wrote:
hello,
I am planning a fairly large hotel VoIP system,
using analog phones. It will
consist of about 100 analog phones, that must have
access to a VoIP server.
I am
when you say support asterisk do you mean IAX only or
sip as well ? I am in a little rush here but I can
write you a pretty big list of SIP providers that are
known to be good.
--- andrew matthews [EMAIL PROTECTED] wrote:
http://connect.voicepulse.net
They support astrisk, with iax2 :)
On
Just realized I should have replied in private and not
to the list. Sorry in advance.
Dovid
--- Dovid Bender [EMAIL PROTECTED] wrote:
I have a half system thats almost done. Had a client
that wanted it then backed out. Please contact me
off
list.
Dovid
--- Michael Sampson [EMAIL
Hehe. This will be the same person looking for a GUI,
not finding one built in to asterisk and cant
understand why.
--- Anthony Rodgers [EMAIL PROTECTED] wrote:
You'll likely find Asterisk itself even more of a
challenge then.
On Feb 15, 2006, at 1:29 PM, roswel ajf wrote:
hi,
Services
[EMAIL PROTECTED]
952-936-4000
Dean Collins wrote:
Whats wrong with the [EMAIL PROTECTED] wake up
configurations?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Wednesday, 15 February 2006
of ATA usage, we had to replace 20% of our
ATAs(overheating,
random dying, coffee, soda, other abuse).
As for initial purchase cost, ATA adapters are
actually slightly
cheaper per port than channelbanks(if buying new).
MATT---
On 2/15/06, Dovid Bender [EMAIL PROTECTED]
wrote:
I may
Because there are cheaper solutions than purchasing 30
gateways that have an RJ11. S/He (sorry abd with
names) would then have to get a channel banker. This
is a lot more costly than some solutions out there.
--- Sam Tam [EMAIL PROTECTED] wrote:
Why not get 30 GSM Gateway from us at £60 each
My GXP-2000 is currently collecting dust. I had
several issues with it. Mainly echo while on speaker.
The other person can barely mae out what you are
saying. Another issue was if the phone recieved to
many calls it would just freeze up and I had to pull
out the plug. Again I have not used it in a
Some people have to stap on others to make them selves
feel good. Very unfortunate.
--- Rusty Dekema [EMAIL PROTECTED] wrote:
I don't think it takes a great leap of the
imagination to infer that
Mr. Kennedy is in fact having the problem he
describes and that,
although it may not be 100%
Don't know about the Dell. I personaly use Cent OS
(www.centos.org) which is RHEL ES without paying for
it. I have it on my server and it seems to be holding
up just fine.
--- Richard OSS [EMAIL PROTECTED] wrote:
Hello,
Digium uses the Dell PE 2850 for their testing.
This site says
We do. It's all about what you say and how you say it.
Making fun of some ones inability to speak english as
well as you do isnt funny. I will not continue
replying to this subject as to not clog the list.
--- Martin Joseph [EMAIL PROTECTED] wrote:
On Feb 19, 2006, at 9:41 AM, Dovid Bender
What kind of services are you looking for ?
--- CyberSource [EMAIL PROTECTED] wrote:
Can anyone recommend a good voip provider? Thanks
___
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best
I personaly use VoipJet, Teliax and myPhoneCompany.
They are all great. Dont remember if teliax supported
IAX. I know that myPhoneCompany for sure dosent. They
use SIP. I did however ind that thier voice quality is
very good.
Can anyone give me some good recommendations for
VoIP providrs that
Marc,
I have a box with two TDM400P's. All of the ports are
FXO's. System is working fine on CentOS.
Regards,
Dovid
Can someone give me a definite answer as to wether
or not you can
reliably run multiple TDM400P's in the same machine?
I need 4 x FXO and 4 x FXS to connect to both the
PSTN
things?
Thanks.
on Monday 02/20/2006 Dovid
Bender([EMAIL PROTECTED]) wrote
I personaly use VoipJet, Teliax and
myPhoneCompany.
They are all great. Dont remember if teliax
supported
IAX. I know that myPhoneCompany for sure dosent.
They
use SIP. I did however ind that thier voice
Again. What do you need ? Incoming and outgoing,
trunking etc. ?
I personaly use.
Voipjet.com
myPhonecompany.com
Teliax.com
I have heard others talk about:
JunctionNetworks
There others that are just not coming to mine. If I
remember them I will try to email them as well.
Dovid
Everything. I
I do not know a lot about centrex but I know that most
PBX's support POTS lines (usually for faxing). You can
have them switch over the lines that they send you to
pots and then you can plug the lines in to a TDM400P.
Regards,
Dovid
--- Devin Heckman [EMAIL PROTECTED] wrote:
Hi,
I'm looking
Peter,
Diffrent companys offer diffrent services. For example
myPhoneCompany offers DID's for both inbound and
outbound. Thier basid DID plan is $5.00 with unlimited
incoming and 60 outgoing minutes. Each additional is
$0.029. Or $10.00 a month with 500 outgoing and the
same rates as above.
Please let us know how it turns out, if there are any
issues etc.
--- Richard OSS [EMAIL PROTECTED] wrote:
Thank you very much. Will go ahead and build the
system. Hope everything goes smoothly.
richard
BJ Weschke [EMAIL PROTECTED] wrote:
On 2/22/06, Richard OSS wrote:
Hello,
I see that you are playing with [EMAIL PROTECTED] How is
it going ? Sorry I have not called you. Been very
busy.
Dovid
--- Tele Cost Price Reducer [EMAIL PROTECTED] wrote:
Manoj,
just look in AMP to Inbound Routing, fill in the
DID, define the softphone
as extension X and send the call to
Ooops. This was meant to be sent direct and not to the
list. Sorry.
Dovid
--- Dovid Bender [EMAIL PROTECTED] wrote:
I see that you are playing with [EMAIL PROTECTED] How
is
it going ? Sorry I have not called you. Been very
busy.
Dovid
--- Tele Cost Price Reducer [EMAIL PROTECTED
Seems that in your iax settings you set the context to
mantra2. You need the same context in extensions.conf
(some one please correct me if I am wrong)
--- [EMAIL PROTECTED] wrote:
Hi All,
I was able to install Asterisk and make outgoing
calls. Recently I purchased two
DID's and I am
This is for the biz list. Please post there.
--- Sam Tam [EMAIL PROTECTED] wrote:
Do anyone know who can provide some cheap PH
routes/.'
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To UNSUBSCRIBE
what do u mean by solution ? please define what you
want.
--- ram [EMAIL PROTECTED] wrote:
Hi
did you got any solution
iam also looking the same solution
if you find kindly tell me which solution works for
u
ram
On 2/27/06, Micke Andersson [EMAIL PROTECTED]
wrote:
did you uncommnet # from before ztdummy ?
--- Sina Bahram [EMAIL PROTECTED] wrote:
Hi all,
I hope everyone is doing well. I just joined the
list, and I've really
enjoyed all I have read about asterisk so far.
Unfortunately, I'm having a
bit of trouble implementing this thing :).
By
I would look at the cost of the channle banks vs.
selling the analog phones and getting very basic voip
hardphones.
--- Conrad Wood [EMAIL PROTECTED] wrote:
Does anyone have any recommendations on how to
connect 160 analogue
phones to an asterisk PBX?
Background information:
A client
I use PICO (nano for CentOS). Works great.
__
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asterisk tends to not work well with mp3's that have
ID3 tags
--- Zach A [EMAIL PROTECTED] wrote:
Hi,
The 3 MP3 files which are installed with asterisk,
what is their bit
rate, are they mono and do they have ID3 tags?
Zach A
___
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
direct - 716.250.3402
mobile - 716.907.4054
email - [EMAIL PROTECTED]
AIM - b2Cory
- Original Message -
From: Dovid Bender [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
direct - 716.250.3402
mobile - 716.907.4054
email - [EMAIL PROTECTED]
AIM - b2Cory
- Original Message -
From: Dovid Bender [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Is the softphone behind NAT ? If it is insert nat=yes
in your dial plan. Is the server behind NAT ? If it is
you need to open ports 5060,5061 and 1-2.
Dovid
--- Leonard Burton [EMAIL PROTECTED] wrote:
HI All,
What is a good tutorial or article on using Xlite to
get into * while
Helo List,
First I would like to apologize for my bad spelling as
well as that I did not search the wiki first. I only
have email access at the moment.
I am having trouble setting both variables and global
variables thru an extension.
I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4
with an
Some one was on my server making changes to my sip.conf files. I am now having trouble with DTMF. No matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I compared it to the wiki and all the configs seem to be in order.Here is my sip.conf
Figured it out. It was simple had to add Answer and hangupDovid Bender [EMAIL PROTECTED] wrote: Helo List,First I would like to apologize for my bad spelling aswell as that I did not search the wiki first. I onlyhave email access at the moment.I am having trouble setting both variables and
Is anyone with a yahoo account having problems
recieving emails from the list. I have not recieved
any emails in about 8 hours and I posted something
about 3 hours ago. If anyone knows please email to
asteriskdigium _AT_ yahoo.com
Thanks
__
Do You
we mirror all the files our selves so our scripts work
flawlessly.
--- Alistair Cunningham [EMAIL PROTECTED]
wrote:
This is a request to the website manager for
asterisk.org.
The build scripts for our ITSP product include the
URLs to download the
Asterisk files, such as:
wget
why not use astcc ? it comes with asterisk and does
all that you have requested. we have scripts running.
one that works via CID and one the user enters the
number.
--- leonimar cape [EMAIL PROTECTED] wrote:
Hi group,
I am currently looking for a prepaid application
that
can do the
already tried it and it didnt work. Could there be any
other files that may have been messed with that is
causing this ?
Dovid
--- Mark Edwards [EMAIL PROTECTED] wrote:
Try dtmfmode=info and see if that works.
Mark
-Original Message-
From: Dovid Bender [mailto:[EMAIL
- Original Message -
From: Ralf Träskman
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Tuesday, January 13, 2009 4:04 PM
Subject: [asterisk-users] 404 not found from one ip-adress
Hi
Our sip provider has two servers that sends calls to our
I think their issue is that they built their business around cheap support
in Asain countries which is a hit or miss. I know that when I pointed out an
obvious flaw that made them look stupid I got email that I had a $20.00
credit with them. I never mentioned it because I did not think it was
You can try blocking the caller ID in the dial plan. Not sure how that will
affect the CDR's. If it does not show up in there in the dial plan you can set
a variable to the caller ID then set it to be blank and on hangup update the
CDR's.
- Original Message -
From: Sriram
To:
I use post variables. I found this on the web. Forgot where I got it from
(sorry that I can't give you credit).
?php
//Connect to the Asterisk Manager
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: username\r\n);
fputs($socket,
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: Wednesday, January 14, 2009 6:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bridge 2 calls
I use post variables. I found this on the web. Forgot where I got it from
Hi,
I am running Asterisk 1.4.22. If in X-Lite I have set to hang-up after 0
seconds of RTP the call gets cut off between 1 1/2 to 2 minutes. I have tried
to connect to another server and the call stayed up. If I take out the (RTP)
setting then it works fine. What would cause the RTP to stop
What is Zap mirroring ?
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, February 09, 2009 10:09 PM
Subject: Re: [asterisk-users] How to make the Asterisk-GUI
Do you have extension ontext 059*162*178*122*78600051 in your
extensions.conf under the default context ?
- Original Message -
From: Philipp Kempgen philipp.kemp...@amooma.de
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Monday, February 02, 2009 10:40 PM
Subject: Re:
I wish we could have this for real
- Original Message -
From: Olle E. Johansson o...@edvina.net
To: Asterisk Non-Commercial Discussion Users Mailing List -
asterisk-users@lists.digium.com
Sent: Wednesday, April 01, 2009 10:18 AM
Subject: [asterisk-users] FOR IMMEDIATE RELEASE: NEW
I just tested. People tend to think that I am a female because of my hi pitched
voice. The system correctly identified me as a male.
- Original Message -
From: Grygoriy Dobrovolskyy
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, February 19, 2009
I had a patch created for 1.4.X for this.
http://bugs.digium.com/bug_view_page.php?bug_id=14159
- Original Message -
From: Gabriel Ortiz Lour
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 18, 2009 8:23 PM
Subject: [asterisk-users] Global h exten
Hi all,
I had a patch created for 1.4.X for this.
http://bugs.digium.com/bug_view_page.php?bug_id=14159
- Original Message -
From: Gabriel Ortiz Lour
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 18, 2009 8:23 PM
Subject: [asterisk-users] Global h exten
Hi all,
How about:
Exten = _X.,3,Dial(SIP/${ext...@carrier,60,M(fax-out))
[macro-fax-out]
exten = s,1,Set(FAXFILE=/root/test.tif)
exten = s,2,Set(LOCALHEADERINFO=WHO CARES WHO I AM ?)
exten = s,3,Set(LOCALSTATIONID=1-800-Who-CARES)
exten = s,4,SendFax(${FAXFILE})
- Original Message -
From:
Stay away form ooh323. It tends to crash Asterisk. If you get h323 let me
know. I was never able to get it working.
- Original Message -
From: bilal ghayyad bilmar...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Saturday, July 11, 2009 19:20
Subject: [asterisk-users] ooh323 and
I know I am responding to an old post but dont think you would want to
change the title of your site from osCommerce to your name ?
- Original Message -
From: Sam Tam [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent:
Hi List,
I have an A200 with echo can. 2-FXO and 2 FXS.
Today I went and upgraded asterisk, zaptel and libpri. I ran the sangoma util to
patch asterisk. When I started up asterisk ZAP1 worked like a charm. However
ZAP2 has been acting up. I only get one way audio on it. The person that I
Anyone here use the Nokia E61 ? I am looking to
invest in a wifi phone and I want to get the best. Is it good as far as
reception ? That is of most importance to me. Thanks.
Dovid
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Doug I have
Exten = 10,hint,SIP/11010
and in mysql I have
exten = 10,1,Dial(SIP/11010)
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, August 21, 2006 3:37 PM
Maybe he is tryin to make it work. This is much better than the old Doug.
Also if he needs features that currently dont exist maybe some one will
create it and then we will all benefit from it :) .
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing
need _both_ in your dialplan.
My extensions.conf has:
exten = 2944054,hint, SIP/2944054
exten = 2944054,1, Dial(SIP/2944054)
ie two lines for the hint.
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 24, 2006 9:32 AM
_both_ in your dialplan.
My extensions.conf has:
exten = 2944054,hint, SIP/2944054
exten = 2944054,1, Dial(SIP/2944054)
ie two lines for the hint.
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 24, 2006 9:32 AM
Message -
From:
Dovid
Bender
To: asterisk-users@lists.digium.com
Sent: Wednesday, August 23, 2006 7:06
PM
Subject: [asterisk-users] One way audion
on Sangoma
Hi List,
I have an A200 with echo can. 2-FXO and 2 FXS.
Today I went and upgraded asterisk, zaptel
and a subsequent asterisk reload.
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Friday, August 25, 2006 7:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime and hints
I dont know why it is working but it is. My
SNOM is a good phone but dosent have QOS. The polycom does :)
- Original Message -
From: Guido Hecken [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 25, 2006 4:52 AM
Subject: RE: [asterisk-users] IP
On 2006-08-24 06:32:27 -0700, Jon Schøpzinsky [EMAIL PROTECTED] said:
I also have this phone, and have stumbled in to the same problem.
I just think that it isn't in nokia's interest to change this, as it
forces consumers to have some sort of local hardware, that (possibly)
only the
of Asterisk's BIGGEST
challenges. We've been stewing over it for a long time.
Doug.
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Friday, August 25, 2006 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime
You can get that on the polycom if you want to fork over another $200.00 +
for the side car. Or if you are using a 601 you can use the first line for
all your calls and then the next 5 for it.
- Original Message -
From: Mario [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On 27 Aug 2006, at 04:03, Dovid Bender wrote:
I was not going to get it based on what people said about the E61
and the NAT issues. Is this false ? I was thinking of getting it
for when I travel to Israel. There seems to be a lot of open wifi
connections all over the country there. Also
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