On Sun, 2005-09-11 at 20:58 -0600, Joseph wrote:
Does anybody runs Asterisk on AMD64?
I can compile it on Gentoo, and start Asterisk a command line but as
soon as I connect any device (like Sipura ATA ), asterisk crashes.
Runs well here.
--
Dave Cotton [EMAIL PROTECTED]
Hi there,
I have a new asterisk working in New Zeland and everything is working
well except when an incoming call to the PSTN hangs up, asterisk wont
hang up the zap trunk (X100P).
I have found this information:
http://bugs.digium.com/bug_view_page.php?bug_id=0001474
Which discusses my
Hi There,
How do i disable the voice mail greeting from an extension?
Thanks
Simon
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On Mon, 2005-09-12 at 19:12 +1200, Simon wrote:
Hi there,
I have a new asterisk working in New Zeland and everything is working
well except when an incoming call to the PSTN hangs up, asterisk wont
hang up the zap trunk (X100P).
With the same type of problem here in France this worked in
Folks,
I have made an inbound call. Now I want to make an out bound.
I have a sip softphone hitting the pbx. On my motorola firewall I
have allowed all traffic in and out to my address. There is not a
firewall on the PBX yet.
here is the log off the softphone..and the PBX logs look like this
Hi Simon!
Try prefixing the MailBox-Number with an 's'.
VoiceMail(s123456)
Simon schrieb:
Hi There,
How do i disable the voice mail greeting from an extension?
Thanks
Simon
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On Mon, 2005-09-12 at 08:51 +0200, Dave Cotton wrote:
On Sun, 2005-09-11 at 20:58 -0600, Joseph wrote:
Does anybody runs Asterisk on AMD64?
I can compile it on Gentoo, and start Asterisk a command line but as
soon as I connect any device (like Sipura ATA ), asterisk crashes.
Runs well
Indead, it's a Siemens DECT Phone (Gigaset) that does not ring.
I could not find the patch, but will start searching in the code.
Thanks for the pointer !!!
Alex
On Wed, 2005-09-07 at 17:46 +0200, Wilson Pickett wrote:
1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it
Dave Cotton wrote:
On Mon, 2005-09-12 at 19:12 +1200, Simon wrote:
Hi there,
I have a new asterisk working in New Zeland and everything is working
well except when an incoming call to the PSTN hangs up, asterisk wont
hang up the zap trunk (X100P).
With the same type of problem here in
On Mon, 2005-09-12 at 20:22 +1200, Simon wrote:
Dave Cotton wrote:
On Mon, 2005-09-12 at 19:12 +1200, Simon wrote:
Hi there,
I have a new asterisk working in New Zeland and everything is working
well except when an incoming call to the PSTN hangs up, asterisk wont
hang up the zap
Hi,
does anybody have a list of VoIP cards wattage?
I'm interested to tdm400p and hfc-s cards (especially BRI beronet cards).
TIA
Giorgio
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Hi,
I recently installed asterisk
1.0.9. Now I want to use the chanspy applicion. Patching and
recompiling is working ok. Loading the application into asterisk is now problem
either. Now I have
the following line in extensions.conf:
exten =
556,1,ChanSpy(-b|scan|Agent)
When an
On Mon, 2005-09-12 at 10:03 +0200, Alex Ongena wrote:
Indead, it's a Siemens DECT Phone (Gigaset) that does not ring.
I could not find the patch, but will start searching in the code.
Thanks for the pointer !!!
Alex
On Wed, 2005-09-07 at 17:46 +0200, Wilson Pickett wrote:
1 phone
Hi,
I have cisco 2600 with fxo card, can we use it to connect to
asteriskathome as SIP trunk..?, so we can use for incoming and outgoing
trunk.
Please need your help.
-mendro-
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On Mon, Sep 12, 2005 at 01:00:45PM +1000, YT Lim wrote:
Does anyone know how to limit syslog file size?
Logrotate only rotates log files (i.e. irrelevant of
file size), and a log file size can grow extremely
large before it is rotated.
Logrotate is invoked (by default) as a daily cron job and
I have a serious problem that repeats very often, after 30 - 50 calls
and I can only solve it by stopped and restarting * :-(
After a while, * seems to loose track of something. When an ISDN call
from PBX needs to go to the Telco, I get
'Ring requested on unconfigured channel 255/255 span 2'
Hi, all
I have managed to connect my * to voipbuster server using IAX. So far, so
good. I have paid them the 5 euro and now my PC aplication has unlimited
local calls. However calls from * are still limited to 1 minute. I have sent
them an e-mail asking about it, but no answer as yet. Could
Hi all,
How can I makea X100P ZAP channel not answering to any incoming calls?
Thanks.
Newbie
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Euh, what is your extensionf.conf part that answers it ?
On Mon, 2005-09-12 at 18:25 +0800, VoIP Newbie wrote:
Hi all,
How can I make a X100P ZAP channel not answering to any incoming
calls?
Thanks.
Newbie
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On Mon, Sep 12, 2005 at 06:25:53PM +0800, VoIP Newbie wrote:
Hi all,
How can I make a X100P ZAP channel not answering to any incoming calls?
Thanks.
And what will it do when a call comes in on the line? nothing (wait for
someone else to answer)? answer and hangup?
The incoming call is
Is this possible to do with the latest asterisk?
thanks
--
Eric Smith
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To
Hi Guys,
I'm using *1.09 with cdr, mysql, add-ons etc on
CentOS with SIP/MyFone in Sydney.
I'm not receiving DTMF from external calls to drive
my menu...
Could someone suggest how to diagnose what/where
the problem may be?
Any suggestions would be most welcome.
Clint.
Dear List,
http://pastebin.ca/22701This is my
problem.
Thanks,
Ozan
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To
I am working with a small inn (under 50 rooms) that is next to a ski
resort. The inn just had Cat5e Homeruns to each room installed, with a
patch panel in the basement. Now it's my job to connect each of the
those rooms to the Internet. I think I have a Cisco switch that I can
do Private
Title: RxFAX
Does anyone have any ideas on why I can fax out using TxFax fine but I can't receive? The system detects a fax and
-- Executing RxFAX(Zap/1-1, /var/spool/asterisk/fax/1124786077.0.tif|debug) in new stack
Slow carrier up
Slow carrier down
Changed from phase 1 to 4
DIS:
Linksys and Netgear switches now also do private VLANs for far less than 6k. They will not provide the features/functionality/management that your 6k Catalyst will provide, but it doesn't sound like you're looking for anything more than making sure traffic from room to room is secure.
While
what is the REGISTER messages are ser saying
Iqbal
Ozan Blotter wrote:
Dear List,
http://pastebin.ca/22701 This is my problem.
Thanks,
Ozan
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Ozan,
Put the following to sip_custom.conf:
[OpenSER]
type=peer
username=8333688231
secret=test
host=212.154.104.198
fromuser=8333688231; some of the following may not be necessary
fromdomain=212.154.104.198
nat=yes
dtmfmode=rfc2833
disallow=all
allow=g729; whichever codec you want
Tzafrir Cohen wrote:
On Mon, Sep 12, 2005 at 01:00:45PM +1000, YT Lim wrote:
Does anyone know how to limit syslog file size?
Logrotate only rotates log files (i.e. irrelevant of
file size), and a log file size can grow extremely
large before it is rotated.
Logrotate is invoked (by
Sorry, I meant:
exten = _0.,1,Dial(SIP/[EMAIL PROTECTED],60,T)
- Original Message -
From: Soner Tari [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 12, 2005 3:25 PM
Subject: Re: [Asterisk-Users]
Hi,
If you are Montrealer (francophone or Anglophone) and interested in talking
about asterisk (usage, business, tips and tricks
), please join us today 6pm
for a coffee/beer at the pub St-Paul in the old Montréal!
See you there,
Adrien
--
Adrien Laurent - CIO
www.modulis.ca
514-284-2020 ext
It is highly unlikely you will get this to work correctly with an X100P
or TDM card due to pci bus issues and missed frames. See the many many
posts relative to this over the last six months or so.
Does anyone have any ideas on why I can fax out using TxFax fine but I
On Mon, 2005-09-12 at 09:56 +0200, Domjan Attila wrote:
On Mon, 2005-09-12 at 08:51 +0200, Dave Cotton wrote:
On Sun, 2005-09-11 at 20:58 -0600, Joseph wrote:
Does anybody runs Asterisk on AMD64?
I can compile it on Gentoo, and start Asterisk a command line but as
soon as I connect
Hi all,
I have installed a TDM22B on an IBM xSeries 220
with FedoraCore 3 and setup asterisk cvs head to work properly.
A few days ago, I tried to update the system kernel
to 2.6.12 from 2.6.9 and also tochange to asterisk stable
1.0.9.
After compiling zaptel and asterisk, i loaded
zaptel
I am working with a small inn (under 50 rooms) that is next
to a ski resort. The inn just had Cat5e Homeruns to each
room installed, with a patch panel in the basement. Now it's
my job to connect each of the those rooms to the Internet. I
think I have a Cisco switch that I can do
On Mon, 2005-09-12 at 16:12 +0300, Dionisis Koumouras wrote:
Hi all,
I have installed a TDM22B on an IBM xSeries 220 with FedoraCore 3 and
setup asterisk cvs head to work properly.
A few days ago, I tried to update the system kernel to 2.6.12 from
2.6.9 and also to change to asterisk
When using the
Comedian mail system, we've set up a remote access trunk to dump to the
VoicemailMain. Problem is, 90% of the time, the system won't recognize the key
tones from the PSTN. anyone else have this problem and end up solving it? We're
using Real Time setup, but the problem can't
anybody succeeded with this issue ? I mean writing xml code to display
queue info on the Cisco screen
jl
2004/12/7, Brian Roy [EMAIL PROTECTED]:
On Sat, 4 Dec 2004 13:08:19 -0600, Joe Dennick [EMAIL PROTECTED] wrote:
I, too would be very interested in this application.
We are also
Makes sense.
Thanks,
TAFF!
--- Flobi [EMAIL PROTECTED] wrote:
If you're talking about asterisk actually doing
anything with the call on a
logical basis (i.e. processing your dialplan),
wait will halt that. Actual
detection (as recorded in CDR) and acknowledment
(via the lower level
Real quick guys, placing a ATA (such as a sipura SPA3K) somewhere similar to
a jack (like on the back of the nightstand where you placed the phone) would
be an easy solution to the problem of using SIP phones. That way customers
can use the normal office type phone or whatever standard phone you
Joseph ha scritto:
On Mon, 2005-09-12 at 09:56 +0200, Domjan Attila wrote:
On Mon, 2005-09-12 at 08:51 +0200, Dave Cotton wrote:
On Sun, 2005-09-11 at 20:58 -0600, Joseph wrote:
Does anybody runs Asterisk on AMD64?
I can compile it on Gentoo, and start Asterisk a command line
Howdy,
1 x TDM400P card with 1 x fxo module.
1 x BT Pots line.
Location - UK
Calls work fine outbound but i'm unable to pickup the
inbound calls.
Asterisk debug:
Asterisk -vvcg
*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing Wait(Zap/1-1, 1) in new stack
--
Yes, I am aware of such switches.
I was trying to stay with something that I could configure and forget.
ALL the linksys gear I have owned, needed to be poked and prodeded
sometimes. IE, Linksys routers stop passing traffic, a simple power cycle
fixes everything. I can't be rebooting the
I have a new asterisk working in New Zeland and everything is working
well except when an incoming call to the PSTN hangs up, asterisk wont
hang up the zap trunk (X100P).
With the same type of problem here in France this worked in zapata.conf.
busydetect=yes ;changed 17.03.04 from
Yeah, I think that would be the right solution.
You could also look at an Epygi Quadro 2x.
It would provide each room with a private lan and one pots line.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Monday,
Small world.
The Inn was going to work on absorbing the cost of the system and the VoIP
service. The phones would be just cheapie grandstream phones, which work
out to about the same as regular analog phones. More features, no cost,
the owners are thinking they can lever this edge to attract
Put on the list the software version that you are using.
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 3252-2977 r
http://www.isolve.com.br
On 09 de set de 2005, at 02:29, Le Van Khoa wrote:
Hi,
I run
On Mon, 2005-09-12 at 15:36 +0200, Massimo De Nadal wrote:
I think you can't run asterisk on a 64 bit linux version. Certainly it
works well on a 64 bit amd processor, but in x86 mode...
Am i right guys ??
Don't think so.
file /usr/sbin/asterisk
/usr/sbin/asterisk: ELF 64-bit LSB
On Sun, Sep 11, 2005 at 11:02:56AM -0700, Derek Whitten wrote:
On Sun, 2005-09-11 at 08:44, Mike M wrote:
On Sat, Sep 10, 2005 at 04:43:26PM -0700, Chris Travers wrote:
Mark Phillips wrote:
The suggestion that I have is for various areas to have dedicated civil
emergency com
What can RAGI do additionally that AGI or FastAgi and DeadAgi cannot do
which is already available under Asterisk?
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of joe
heitzeberg
Sent: Sunday, September 11, 2005 12:31 PM
To: Asterisk Users
No dialpad key buttons seem to work dialing out.
Latest CVS head.
Sep 12 09:41:55 DEBUG[1941] chan_zap.c: Exception on 15, channel 1
Sep 12 09:41:55 DEBUG[1941] chan_zap.c: Got event On hook(1) on channel 1
(index 0)
Sep 12 09:41:55 DEBUG[1941] chan_zap.c: disabled echo cancellation on
channel
I am on UK Cable (Telewest) will this callerid patch work with the
cable caller id?
Thanks.
Paul.
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On Sun, Sep 11, 2005 at 02:45:47PM -0700, Michael D Schelin wrote:
BUT,
let me tell you about how bad the southern CA. radio site owners are
becoming. We had a 4 day outage at a very large site where one of my
radios is located. None of them care anymore about backup power. This
happened
Since I can't seem
to get anything figured out for the Comedian system, are there any other systems
out there that we can hook asterisk into?
Sherwood McGowan
ViaTalk
Level 2 Support
VOIP System Engineer
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I have several installations running Asterisk over FC3 x86_64 on Tyan
Opteron motherboards. I also have one installation on an Athlon 64.
When compiling zaptel, be sure to 'make linux26' for the 2.6 kernel.
Massimo De Nadal wrote:
Joseph ha scritto:
On Mon, 2005-09-12 at 09:56 +0200,
I haven't seen your other posts. What exactly can't be figured out?
Sherwood McGowan wrote:
Since I can't seem to get anything figured out for the Comedian
system, are there any other systems out there that we can hook
asterisk into?
Sherwood McGowan
ViaTalk
Level 2 Support
VOIP System
On Mon, 2005-09-12 at 09:12 -0600, Michael Welter wrote:
I have several installations running Asterisk over FC3 x86_64 on Tyan
Opteron motherboards. I also have one installation on an Athlon 64.
When compiling zaptel, be sure to 'make linux26' for the 2.6 kernel.
From certainly 1.0.9 even
I have a problem with the VoiceMailMain not catching the username and
password of a user when they use our external number to access voicemail.
The extensions.conf file looks at incoming calls, finds that the call came
into our toll free number for voicemail and then passes the customer to
I have 1 x100p. Caller id works fine with the beta1 release. Cvshead
releases fail with a combination of checksum and ss_thread errors?
I'm concerned when beta2 or the 1.2 release comes out it will not work.
I have been through the configs I can't find and changes that need to be
made to get
---BeginMessage---
Hy,
I've bought a swissvoice IP10s. I've configurated asterisk for a peer
and user with this telephone. The telephone is registered correctly in
asterisk but I can't do call with the phone. I have read that the
principal reason for this problem is the firmware. I don't
Beloware what I have in extension.conf.
exten = s,1,Goto(1234,1)exten = s,2,Hangup
exten = 1234,1,Dial(SIP/1234,30,t)exten = 1234, 2, hangup
I would like to have channel 1 answer incoming calls and channel 2 not answer any incoming call (outgoing calls only).
1. I don't know how to
Hi
I have been playing with chanspy, to see if I can get it to work with
particulat extensions, within my virtual pbx setup
I have tried
exten = ,1,ChanSpy(scan,SIP/1234) to listen to anything matching
1234, but since my phones are not registered with asterisk, but in SER,
my sip
I found everything I needed to know as comments in the config files.
First test I did was to setup a few boxes for friends and family to make
outbound calls. They love it. They can place a call from cell phone and
the outgoing caller ID is what they want the caller to see. I also like
using it
Hello All
any body have used SS7 run with asterisk. could you like tell me how to download driver of SS7 and how to use it.
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IMHO, I would imagine that you would be best advised to use the
existing analog phones and wiring with a channel bank or two, and
then use the cat5 cable for internet. I think it might even be
possible to make the MWI on the analog hotel phones work with Asterisk.
Billing, however, is a
Have you seen the 3Com LAN-switch-in-a-wall-jack device? It's a four
port device that could also be used for the guest's PC. It can supply
PoE to the phone using a wall wart or PoE on the incoming LAN circuit.
You'll need some logic between Asterisk's management interace and the
property
Some extra info about this strange thing:
- even with CLI pri intense debug span 1...4 on, I get NO extra
debugging. (On the CLI, I see only 'Ring requested on unconfigured
channel 255/255 span 2'
- When I 'redial' on the Phone (connected to the PBX) several times,
I sometimes get through
I love Comedian Mail. The problem is only that I can't get the VoiceMailMain
to take the users password and number when they get passed to it from the
external access number. I've seen other people with this same problem
posting online in various places, and I'm trying to find anyone that got a
Hi all,
how may I set in callfiles additional variables so that I might reference
them later in the extensions.conf ?
eg. TESTDATA=ABCD
I didnt find an example in www.voip-info.org.
Is it possible at all ? I ask because I need several vars beyond
ACCOUNTCODE.
Are there difference in the
On Mon, 2005-09-12 at 23:34 +0800, VoIP Newbie wrote:
Below are what I have in extension.conf.
Is this the complete file ?
exten = s,1,Goto(1234,1)
^^^
is used to jump to contexts. ?
exten = s,2,Hangup
exten = 1234,1,Dial(SIP/1234,30,t)
He's just doing one shop though so all of those things may not be a
requirement.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Welter
Sent: Monday, September 12, 2005 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Paul Goodyear wrote:
I am on UK Cable (Telewest) will this callerid patch work with the
cable caller id?
It's probably not needed. Cable companies in the UK use a different
form of callerid which is closer to bell than the BT format.
Depending on your cable company/area you may be using
You can add your library path in the /etc/init.d/asterisk script and
restart asterisk via service.
or
You also can
1/ add your library path in your /etc/ld.so.conf
2/ type ldconfig as root.
3/ restart asterisk via service
Hi List,
My AGI seems work well in
A lot of that could be accomplished with the TAPI driver for Asterisk.
It's easy to write a simple application for the TAPI and even Outlook will use
it.
Sometime I plan to develop a TAPI program for Asterisk, but I haven't had the
time recently.
Regards,
Chris
- Original
Hi Jon,
perfect instructions, thank you very much.
I used them and after some tests they worked.
Here is what I did. Maybe someone could use it completely:
Install helpers:
apt-get install kernel-package ncurses-dev fakeroot wget
Massimo De Nadal ha scritto:
I think you can't run asterisk on a 64 bit linux version. Certainly it
works well on a 64 bit amd processor, but in x86 mode...
Am i right guys ??
It does work all right here as a native 64-bit app, on a 64-bit kernel
with zaptel and BRIstuff!
Greetings,
--
Mike M wrote:
On Sun, Sep 11, 2005 at 02:45:47PM -0700, Michael D Schelin wrote:
BUT,
let me tell you about how bad the southern CA. radio site owners are
becoming. We had a 4 day outage at a very large site where one of my
radios is located. None of them care anymore about backup power.
Hi
I'm trying to build an Asterisk packages for a C3 system (256MB memory,
cpuinfo below).
/proc/cpuinfo:
processor : 0
vendor_id : CentaurHauls
cpu family : 6
model : 9
model name : VIA Nehemiah
stepping: 8
cpu MHz : 1000.736
cache size : 64
Lots of folks have this working, so you've got something messed up in
your config which 'could' include inadequate gain settings if your
using an analog pstn interface, dtmf settings, etc.
Give us a clue what you actually have installed for a pstn interface
along with the conf file entries that
I don't get this?
Is included on 1.0.9 or is not? I know that a lot of people was trying it,
but just to be clear, is T.38 passthrough included on 1.0.9?
Thanks,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent:
Good news for service providers in my opinion, Ebay will likely start
alienating Skype users like they did with Paypal users.
SkypeSucks.com domain already taken, shucks
http://www.kesq.com/Global/story.asp?S=3837895
--
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com -
Are there any measureable advantages to using Asterisk on X86_64 Linux (CPU
utilization, realtime performance, echo can, transcoding etc.) ? What about
disadvantages (memory usage, thunking, etc.)
-Original Message-
From: Emanuele Pucciarelli [mailto:[EMAIL PROTECTED]
Sent: Monday,
Is there a way to add DNIS digits, and pass it onto another line?
My provider will noy supply DNIS digits over analog lines, so I'd like
to take a call on a trunk, add some DNIS digits, and pass it to another
asterisk system...
-Darren
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--Bandwidth
Hi,
We run Asterisk/Zaptel 1.0.5-BRIstuffed-0.2.0-RC7i on an Intel Celeron (1.4
GHz). The hardware setup contains - besides of a Wildcard TDM400 Card with
4 FXS modules and a Wildcard TE410P - an additional TV/video card
(Hauppauge PVR-350).
Now we experience the following behaviour: As
This is the most insane decision taken by eBay ever.
If I was an investor I would be devastated.
There is nothing wrong with a company starting up a voice over ip
service but this isn't what eBay should be doing.
Companies need to understand they are stewards for their investors money
not the
I have no pstn interface, I'm running SIP only system. I get pstn
connections via a provider that passes the calls to me via SIP. All my other
menus work just fine, so I don't see a problem there, unless comedian mail
is just more sensitive.
Thanks
--Original Message-
-From: [EMAIL
As an ex 7th employee of PayPal before it was sold to ebay and having
gone through Ebay's carenot attitude to it's customers, let alone
investors, I concur with Dean's comments on ebay's future.
If you still have ebay's shares in your closet, get them out now.
Seshu Kanuri
Dean Collins Wrote:
Whats wrong with dial(zap/channel/${EXTEN}addeddnisdigitsgoeshere)
On 9/12/05, Darren Wright [EMAIL PROTECTED] wrote:
Is there a way to add DNIS digits, and pass it onto another line?
My provider will noy supply DNIS digits over analog lines, so I'd like
to take a call on a trunk, add some
While ostensibly people see this purchase as Ebay being a me-too VoIP
player with GoogleTalk and Microsoft buying, what's it called, Teleo? , I
think there's a deeper plan. Let's look at the numbers:
-Huge installed base, like 30 to 70 million users (those are the numbers I
hear kicked around)
What about speeding up the time it takes to have an auction?
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Monday, September 12, 2005 1:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Skype purchased by Ebay 2.6
Hi
70 million users, now how many of these are ALREADY ebay customers.
Google never made a succes out of any other thing other than search and
that will remain the case, companies never do, they are usually good at
what they started at, especially if they grow very big, then they
decided
Efforts are underway to setup a New York Asterisk User Group which is
free, and open to the public.
We're currently looking for members who share a passion for creating,
and discussing, interactive telecommunication services with Asterisk
and other Open Source technologies.
* Meeting
That theory sounds great on paper, but consider some of the factors
below.
1) On the web there are no loyalties. Today it is Skype but tomorrow
certainly belongs to GoogleTalk, if they can put Encryption. How
difficult is that to do? Whose quality is better? Skype or Googletalk? I
read that
Hi Seshu,
RAGI communicates with your Asterisk server over a socket, allowing
you to create your call handling scripts in Ruby or Ruby on Rails.
This allows you to build complex routines that using object models and
database lookups in Ruby [on Rails], and not have to split your
business logic
On Mon, Sep 12, 2005 at 06:26:31PM +0200, Rainer Maier wrote:
Hi Jon,
perfect instructions, thank you very much.
I used them and after some tests they worked.
Here is what I did. Maybe someone could use it completely:
Install
Hi Mohammad,
We have created RAGI - ruby asterisk gateway interface as a means of
bridging Asterisk with Ruby or Ruby on Rails.
http://ragi.sourceforge.net/
We have a set of things including:
+ object to encapsulate AGI
+ server object to embed in web applications framework (e.g. Rails) or
as
What about setcallerid() I don't remember the new syntax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, September 12, 2005 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Adding DNIS
[EMAIL PROTECTED] wrote:
Hi all,
how may I set in callfiles additional variables so that I might reference
them later in the extensions.conf ?
eg. TESTDATA=ABCD
I didnt find an example in www.voip-info.org.
Is it possible at all ? I ask because I need several vars beyond
ACCOUNTCODE.
Rumour has it, that C3 performance is poor. You don't expect anything
but low power, small form factor OpenOffice running xface on C3. C3 is
cheap and low power.
On 12/09/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Hi
I'm trying to build an Asterisk packages for a C3 system (256MB memory,
Seems to me that the intellectual property (the network + encryption +
client) is really the crown jewel here and an Ebay/Skype client is a no
brainer. In fact, I can see a scenario where they drop or severly
de-ephasize the voice part for their ends - maybe reduced to a Powered by
Skype blurb on
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