Hi,
There's another interesting approach to filtering SIP scanners that I
found on the DSLReports VoIP forum:
http://www.dslreports.com/forum/r29375858-SIP-Scanners
Basically allow known IP addresses in your iptables rules, and for the
rest of the world, allow only clients that use a specific
as they come from the same IP
address/port as the previous registration for the same peer.
Thanks,
Liviu Toma
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You can't have two web servers at the same IP address using the same
port. Either you'll have to use different ports, or use only one web
server to host multiple web sites.
On Thu, Feb 9, 2012 at 1:57 PM, Henry Coleman henry.cole...@voip-pbx.ca wrote:
I want to build a web/email server to
Almost any modem can work in bridge mode and you can use another
router to establish the PPPoE connection and do the routing. Even with
the Bell modem/router combos (including the 2wire 2701 which they uses
these days), you can factory reset it and don't configure anything,
then connect another
Hello,
One of my asterisk servers is in Mississauga connected to a PRI line
from Telus. Since Bell/Telus still think we're in the 19th century and
calls from Mississauga to for example Markham are supposed to be long
distance, I am looking to build a way to handle local/long distance
exchanges
My vote (in this order):
East of Toronto
Downtown
North York
On Thu, Sep 15, 2011 at 11:19 AM, Stephan Monette
monette.step...@gmail.com wrote:
Hi everyone,
I'm currently working on organizing our next TAUG meeting and looks like we
will be able to have the meeting sometime during the week
Thanks for taking the initiative on this, Stephan. It's definitely a
good idea to have those meetings again.
Liviu
On Fri, Aug 19, 2011 at 10:13 AM, monettes smone...@primustel.ca wrote:
Hi everyone,
It's been a while since we had our last meeting and I though we could
organize a meeting in
Hello,
I am looking for some advice on DECT IP phones.
The company I work for is planning to swtich the current PBX to an
Asterisk based PBX and IP phones. I don't like the idea of ATA+regular
phone for an office environment and we need cordless phones, so I am
looking at one of these:
- Aastra
400, Panasonic KXTPA50, snom M9, snom m3
On 14 April 2011 10:21, Liviu Toma liviu.t...@gmail.com wrote:
Hello,
I am looking for some advice on DECT IP phones.
The company I work for is planning to swtich the current PBX to an
Asterisk based PBX and IP phones. I don't like the idea of ATA
Here's a few things that I do and it always works for me:
- use a STUN server under the SIP tab (any public server will do, for
example stun.softjoys.com), set Substitute VIA Addr: to yes and STUN Enable:
to yes
- change the default registration time under Line 1 and Line 2 from one hour
to
Is this for residential or business ?
Any regular Bell copper line will wok for fax. As far as I know, the
residential is about $25 / month.
Liviu
On Mon, Dec 6, 2010 at 10:10 AM, TAUG subscriber t...@fubutel.net wrote:
Hey I just asked Bell and they claim that a fax line is $49.02/mo or $36.43
My bad, it's about $3 more with the touch tone service and 911 fee.
Liviu
On Mon, Dec 6, 2010 at 10:17 AM, Liviu Toma liviu.t...@gmail.com wrote:
Is this for residential or business ?
Any regular Bell copper line will wok for fax. As far as I know, the
residential is about $25 / month
on the commit time
from Bell. Just thought someone knew of another way (other than claiming
residential).
Thanks anyway.
Erik.
-Original Message-
From: Liviu Toma [mailto:liviu.t...@gmail.com]
Sent: Monday, December 06, 2010 10:21 AM
To: asterisk@uc.org
Subject: Re: [on-asterisk] Bell
I have a couple of these as well, installed behind NAT firewalls and
talking over the internet to my hosted asterisk (on a hosted server,
public IP) and I had no issues. Also video calls are working fine in
this scenario.
Liviu
On Thu, Nov 18, 2010 at 7:34 AM, Chris Chen chris.chen2...@gmail.com
appreciate
some technical feedback here - to get it rolling.
*
*
*Thanks and regards,*
*Reza.*
On Thu, Nov 18, 2010 at 7:47 AM, Liviu Toma liviu.t...@gmail.com wrote:
I have a couple of these as well, installed behind NAT firewalls and
talking over the internet to my hosted asterisk
Hello,
Has anyone tried Asterisk 1.8 yet ? Does it work with config files
from 1.6.2 or it requires some (or a lot of) changes ?
Thanks,
Liviu
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To unsubscribe, e-mail: asterisk-unsubscr...@uc.org
For additional commands,
Don't waste your time with the username and password you use with the
softphone. For the SIP settings, they are using the phone number with a 1
in front as username, and an 8 character randomly generated password.
The server used by the softphone is the same as the one they give you for
the ATA:
One thing nobody has mentioned was which network was the 3G stick from.
I doubt all carriers have the exact same coverage.
Liviu
On Mon, Aug 2, 2010 at 8:16 AM, Frank Bax f...@sympatico.ca wrote:
Andrew Kohlsmith (mailing lists account) wrote:
On Sunday, August 01, 2010 02:03:07 pm Bruce N
The SPA1001 supports two accounts on a single phone jack (select
between them using the # key).
The SPA3102 supports one account with registration (for inbound and
outbound) and I think up to 4 additional accounts for outbound only
(no SIP registration). The dial plan can be used to select between
There's also the 365 and 742 area codes reserved as overlay for 905
(starting March 2013). Not sure why, they barely started assigning
numbers from the 289 area code.
On Tue, Jun 1, 2010 at 1:00 PM, Dave Donovan donovan.da...@gmail.com wrote:
I'm sure many of you know about this already but it
Other great remote access devices that can connect over the serial
port are the Livingston PortMaster boxes. Those are suitable
especially when you need to control multiple servers (some of the PM
boxes have 24 serial ports or more). They can be purchased on eBay for
less than $100
On Tue, Jun 1,
The SPA won't give a dial tone if it's not connected to the network.
If you have a router around that can give it an IP address, that would
do the trick (even if the router's WAN is disconnected).
Also, if you dial a digit, the SPA will go silent but after a few
seconds, if nothing else happens,
Hello,
Does anyone have a sample DAHDI configuration for a Digium TE110P card
? I need both the /etc/dahdi/system.conf and
/etc/asterisk/chan_dahdi.conf
I have mine half working, as in the PRI is up, the D channel comes up,
when I make an inbound call Asterisk answers it, but as soon as it
starts
Thanks, I had some extra settings under [trunkgroups] in
/etc/asterisk/chan_dahdi.conf which were causing the issues. I removed
them and the PRI works OK now.
Thanks,
Liviu
On Wed, Apr 28, 2010 at 3:45 PM, Dave Donovan donovan.da...@gmail.com wrote:
On Wed, Apr 28, 2010 at 2:57 PM, Liviu Toma
Has anyone used a Cisco phone with the SIP firmware connected to an
Asterisk in a remote location (over internet) ?
My Cisco phone is a modest 7905 but I had a lot of issues with it
because it doesn't seem to have any kind of keep-alive mechanism or
STUN support like the SPA series have.
Liviu
version of the
firmware, its terribly buggy.
I am assuming you are running your remote phones via vpn of some kind.
-Original Message-
From: Liviu Toma [mailto:liviu.t...@gmail.com]
Sent: Friday, April 09, 2010 1:50 PM
To: asterisk@uc.org
Subject: Re: [on-asterisk] Re: Cisco 7940
A little bit off topic, but check today's home page at http://www.kernel.org :)
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When I saw the thread about virtualization a few weeks, I got really
excited about Xen and the PCI pass through feature (I was planning to
use a Sangoma T1/PRI card). So, in my ignorance, I put together a
Citrix XenServer and installed asterisk 1.6.2.6 on a virtual machine.
To my disappointment, I
I've recently unlocked two Mediatrix 2102 ATAs and I am trying to set
them up with my Asterisk but I am not having much luck.
I used their Unit Manager to set them up, and the only options I
changed are the the SIP registrar, proxy and SIP accounts but it
doesn't even attempt to register with my
, in the Authentication settings,
Validate Realm needs to be turned off.
Liviu
On Sat, Mar 6, 2010 at 10:01 AM, Liviu Toma liviu.t...@gmail.com wrote:
I've recently unlocked two Mediatrix 2102 ATAs and I am trying to set
them up with my Asterisk but I am not having much luck.
I used their Unit Manager to set
Maybe it needs a nat=yes in the [general] section of sip.conf ?
On Mon, Mar 1, 2010 at 12:01 PM, Bruce N het...@hotmail.com wrote:
Is there any other place than sip.conf and any other settings other than
externip=x.x.x.x that would effect the sip packet? I can clearly see that
sip packets
Quintum AFM800 may work for you. Has 4FXS, 4FXO and support for NAT
Liviu
On Tue, Feb 23, 2010 at 3:20 PM, birchstr...@gmail.com wrote:
Hi there, can anyone recommend an ATA similar to the Audio Codes MediaPack
118 analog VoIP gateway, 4FXS, 4FXO that has built in NAT support? Key is
NAT
The Audiocodes gateways I have seem to support STUN, isn't that what
you're looking for ?
On Tue, Feb 23, 2010 at 3:20 PM, birchstr...@gmail.com wrote:
Hi there, can anyone recommend an ATA similar to the Audio Codes MediaPack
118 analog VoIP gateway, 4FXS, 4FXO that has built in NAT support?
SPA8800 has 4 FXS and 4 FXO
http://www.cisco.com/en/US/products/ps10446/index.html
On Tue, Feb 23, 2010 at 5:31 PM, John Lange j...@johnlange.ca wrote:
On Tue, 2010-02-23 at 15:20 -0500, birchstr...@gmail.com wrote:
Hi there, can anyone recommend an ATA similar to the Audio Codes MediaPack
See if the bank terminal can be configured to dial a prefix before the
actual phone number. If it can, make it dial *99. This is the Modem
Line Toggle Code, which is valid for all Linksys ATAs and basically
prepares the ATA for a modem call: it forces the G711 codec, turns
off echo canceling, call
This is what works for me: the useragent is set to just asterisk (no
quotes) and the entries in sip.conf are like below. Asterisk 1.6.2.1
right now, but I've been using it since 1.6.0.x
register = 1416...@fpl_peer
[fpl_peer]
type=friend
context=default ; the default context for incoming
you can make. Or, if you have the time, you
can call me at SIP:1...@snowbirdphones.com, ext 1.
Thanks very much,
--terry
On Fri, Feb 12, 2010 at 02:22:08PM -0500, Liviu Toma wrote:
This is what works for me: the useragent is set to just asterisk (no
quotes) and the entries in sip.conf
To access the Provisioning settings of the PAP2T, use 110# to find
its IP address, then go to the web interface, click Admin Login on the
right then click Switch to Advanced view in the middle. Now you should
see the Provisioning tab (see this for what the web page should look
like:
Sipuras. Only reason I'm using those instead of the PAP2's is T38...
although I have yet to have to actually use T38.
-Original Message-
From: Liviu Toma [mailto:liviu.t...@gmail.com]
Sent: January-29-10 9:15 AM
To: asterisk@uc.org
Subject: Re: [on-asterisk] Help with auto provisioning
I know someone who had the same issue and they managed to get it zeroed, but
the situation was a little different.
The LD company was Bell. The problem was that Bell were not supposed to be
the LD. Basically the company moved from analog lines (from Bell) to a PRI
(from Bell too). The LD for
at 10:37 PM, Patrick Song stl...@gmail.com wrote:
I had a bad luck to send sip traffice over Rogers Portable Internet in the
past but IAX is ok
On Wed, Jan 27, 2010 at 7:44 PM, Liviu Toma liviu.t...@gmail.com wrote:
Hello,
Has anyone been able use a VoIP adapter over WiMAX (Bell Rural
Hello,
Has anyone been able use a VoIP adapter over WiMAX (Bell Rural Internet or
Rogers Portable Interet) ?
I have a Bell WiMAX modem to play with for a couple of days and I can't get
VoIP to work through it.
Basically the adapter registers with Asterisk or with the other service
without
, Liviu Toma wrote:
Actually the Asterisk logs show nothing and there is no mail log.
While I was looking around I found a file called astmail-5I3nSR in /tmp/
I tried running the sendmail command exactly as in the mailcmd=
statement in voicemail.conf passing the file in /tmp and I got the
email
.
On 2010-01-03, at 8:54 PM, Liviu Toma wrote:
I already did that, of course, because the utility is called
mini_sendmail and is located in /opt/usr/bin
Thanks,
Liviu
On Sun, Jan 3, 2010 at 8:46 PM, Andre Courchesne
courc...@net-forces.com wrote:
Ok, then you probably need to change
The Sipura Profile Compiler can generate the default XML for you.
The SPC tool is included with the newest firmware, which you can get
from here
http://www.cisco.com/en/US/prod/voicesw/ps6790/gatecont/ps10024/ps10026/SPA2102_Firmware.zip
Here's the command line to generate the sample XML file:
One thing I forgot to mention: the sample XML generated by the SPC
tool is quite huge and you really don't need to change all the options
you see there. I attached a stripped down XML which I normally use to
provision my adapters.
Liviu
On Wed, Dec 16, 2009 at 10:53 AM, Liviu Toma liviu.t
I'd say the most popular ones for residential use are vbuzzer.com,
acanac.com and lately, freephoneline.ca
Liviu
On Thu, Dec 10, 2009 at 10:34 AM, Ivan Avery Frey ivan.f...@utoronto.ca wrote:
Is there a current list of Toronto VOIP providers? Any providers you
recommend? I'm setting my friend
:
This may be the issue you're experiencing. It will be resolved in the
next non-security release of 1.6.1.
https://issues.asterisk.org/view.php?id=14828
On Mon, Nov 2, 2009 at 10:20 AM, Liviu Toma liviu.t...@gmail.com wrote:
Hello,
Does anyone use Asterisk 1.6.1.6 or higher
Hello,
Does anyone use Asterisk 1.6.1.6 or higher with a voicenetwork.ca account ?
When I upgrade mine to 1.6.1.6 or 1.6.1.8, the caller no longer hears a ring
tone when they call in, and when I pick up the call there's no sound going
either way. If I go back to 1.6.1.4 everything is fine.
All
Replace the dial plan string with this
(S0:4165551234)
where 4165551234 is the number that will be dialed automatically
Liviu
On Thu, Oct 22, 2009 at 12:59 PM, RICHARD KWOK rtck...@rogers.com wrote:
Hi
I was told that PAP2T can be setup to work in the off-hook
redial mode so that when the
I have the same type of plan. I set up call forwarding when not
reachable / no answer to go to a freephoneline.ca account and the
freephoneline account is set to e-mail me the voicemails.
Liviu
On Thu, Oct 15, 2009 at 8:02 AM, Bruce zhang bruc...@gmail.com wrote:
Hi Guys,
Maybe this is not
*67 should disable it permanently (until you enable it back with *68), but
you can add it automatically by using the dial plan. For example, let's say
your dial plan is something like
(*xx|911|[2-9]x|1[2-9]x|0.)
Let's say want to prepend *67 in front of the local and long
- Original Message -
From: Mike Ashton
To: Liviu Toma
Sent: Sunday, June 28, 2009 8:06 AM
Subject: Re: [on-asterisk] Re: Making Asterisk listen on 5060 and 5061
Liviu,
The easier solution is to configure the ATA to use 5060 5061, and both
register with your asterisk server on port 5060
Hello,
Is there any way to make Asterisk listen on two UDP ports for SIP messages,
such as 5060 and 5061 ?
I am using an ATA that has 2 lines (Motorola VT2142, but it applies to other
adapters as well) and I am trying to make both register with Asterisk. The
problem is both channels send
I forgot to mention: when both channels of the ATA are enabled, neither one
is able to register. If I disable either one of the channels, the other one
registers right away, so I am guessing all the settings should be OK.
Liviu
- Original Message -
From: Liviu Toma liviu.t
For a cheaper DD-WRT solution, you can use WRT54G2 which can be
flashed with the micro build of DD-WRT.
-Original Message-
From: Apache [mailto:apa...@tsx3.computeradvocacy.com] On Behalf Of Henry
L.Coleman
Sent: Friday, June 12, 2009 11:35 AM
To: asterisk@uc.org
Subject: Re:
Almost any VoIP phone or ATA supports remote provisioning. Basically you can
set it to download its configuration file from a HTTP or TFTP server every X
seconds. I do this with all my Linksys ATAs, which I set to download the
configuration every 30 minutes. When I need to make a change, I change
If the OS is CentOS, Fedora or something else that resembles RedHat,
the config files for the NIC cards contain the MAC address of the
card. The config files are called something like ifcfg-eth0 and are
located under /etc/sysconfig/network-scripts You can simply comment
out the MAC address or
My home dry loop (in Toronto) has dial tone but I can only dial some
Bell numbers (I believe they are called ANAC) which will automatically
read back the number for my dry loop line (yes, in Toronto, every dry
loop seems to have a number attached to it, which, when called, comes
up with a message
I have my asterisk running in a datacentre with a public IP address, and a
whole bunch of Linksys ATAs behind a PFsense firewall in another location,
connected to that Asterisk.
I have never experienced any issues with them. All the Asterisk peers are
configured with nat=yes, and the Linksys
For vbuzzer, you must have an extension in the format
exten = yourvbuzzerusername,
in your extensions.conf, in the context associated with your vbuzzer peer.
That's where your incoming calls will land
I don't know about voipgo, I haven't used them
Liviu
- Original Message -
From:
Ncix.com seems to carry an Intel board with dual core Atom:
http://www.ncix.com/products/productdetail2.php?noheader=1sku=33303
They also have it as a bundle (case+mobo)
On Wed, Jan 21, 2009 at 11:11 PM, Chuck Mariotti cmario...@xunity.comwrote:
Stephan,
I just wanted to find out where you
That's usually a feature of the phone or the ATA (if you're using a
regular phone connected to te ATA).
I have only done this with Linksys ATAs and it's part of the dial
plan. For more info see the admin guide, chapter 3-5:
http://www.provu.co.uk/pdf/linksys/Linksys_ATA_ADMINGUIDE.pdf
Liviu
On
-
From: Liviu Toma [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 11, 2008 11:22 AM
To: Remzi Semsettin Turer
Cc: TAUG
Subject: Re: [on-asterisk] Asterisk and STUN
Thanks very much for the reply, however those articles show how to set
up a stun server. What I am looking for is how to set up
Can an Asterisk server located behind a NAT router use an external
STUN service to find its public IP address (externip or externhost) ?
Thanks,
Liviu
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To unsubscribe, e-mail: [EMAIL PROTECTED]
For additional commands, e-mail:
Sure, but it may be helpful to give us the range (or just the digits
that are different at the end)
Let's say the range was 441234567899 through 441234567999
then your dialplan will be
exten = 441234567899,1,Goto(a2billing-did,${EXTEN},1)
exten = _4412345679XX,1,Goto(a2billing-did,${EXTEN},1) ;
you said?Thank you,Tudor
--- On Tue, 10/28/08, Liviu Toma [EMAIL PROTECTED] wrote:
From: Liviu Toma [EMAIL PROTECTED]
Subject: Re: [on-asterisk] Dialplan question
To: [EMAIL PROTECTED]
Cc: Andre Courchesne - Consultant [EMAIL PROTECTED], Simon P
Hello,
I would like to implement something like the *69 feature in my Asterisk (I
know that some adapters (Linksys and Sipura at least) handle it by
themselves, by storing the number of the last caller and redialing that
number when you *69 but I have some ATAs which don't have the feature and
you can use the local
channel:
exten = *69,1,Dial(LOCAL/[EMAIL PROTECTED])
Cheers,
spd
On Mon, 4 Aug 2008, Liviu Toma wrote:
Hello,
I would like to implement something like the *69 feature in my Asterisk (I
know that some adapters (Linksys and Sipura at least) handle it by
themselves
It doesn't. There's no dialtone at all.
On Thu, Jun 26, 2008 at 8:47 AM, Simon P. Ditner [EMAIL PROTECTED] wrote:
Does dryloop DSL have 911 service on it, or is the line truly phone
service free?
-
To unsubscribe, e-mail:
I've been trying a lot of stuff about freephoneline but didn't have
much luck (I tried both on asterisk and a couple ATAs). I think they
need to flag the account as capable of working with a device other
than their softphone, otherwise it won't work.
Not sure if it helps, but some of the facts
Using Asterisk 1.4 with a Sangoma PRI Card (A101). Is there anything
particular that I need to setup in order to send/receive the Caller ID with
name ? (assuming the switch I am connected to supports it).
Currently my card seems to be sending and receiving only the number part of
the Caller ID.
Hello,
Does anyone have the latest firmware for the Cisco 7912G phone ? Or at
least the cfgfmt utility and a template config file.
I don't have a Cisco SmartNet cotract to download it from their site :(
Liviu
-
To unsubscribe,
As for GeoIP, doesn't tend to work very well with DSL, especially
where the tails are terminated 100's or 1000's of km from the actual
customer
I can confirm that. I live in Toronto, but most GeoDNS service locate me in
Chattam, ON because my DSL internet service is from a company located there
Personally, I wouldn't pay a dime for any USB phone. Technically from what I
understand they are pretty much just a sound card. The actual SIP or IAX
stack is still implemented in a software application installed on the PC, so
you're better off with just a good headset and a softphone like
Yeah, mine too seems to ring somewhere (I get a ring tone when I call it),
but it's not coming to my Asterisk.
Liviu
- Original Message -
From: David LEWIS [EMAIL PROTECTED]
To: TAUG asterisk@uc.org
Sent: Wednesday, April 16, 2008 4:17 PM
Subject: [on-asterisk] Anybody have DID's with
Hello,
I am looking for a way to place a call autmatically to burn minutes on an
inbound VoIP account (one of thise free accounts hat has to be used at least
once every X days, otherwise it expires). I can place the call by placing a
file /var/spool/asterisk/outgoing from a crontab job, I can
(someaudiosothelineisnotquiet)
exten = s,n,Hangup()
Then just have the destination channel for the call file be
Local/[EMAIL PROTECTED]
Jim
Liviu Toma wrote:
Hello,
I am looking for a way to place a call autmatically to burn
minutes on an inbound VoIP account (one of thise free
accounts hat has
The specs for HT503 look very similar to HT488. Any idea what's the
difference between the twe, except for design, connectors, layout etc
?
On Fri, Mar 28, 2008 at 10:58 AM, Andrew Kohlsmith (lists)
[EMAIL PROTECTED] wrote:
On March 28, 2008 10:46:22 am Dave Sullivan wrote:
I've got a cheap
I've been looking at allo.com and I am a bit skeptical:
- there's nothing on their web site or checkout page about shipping charges
- nobody answers the toll free number
- the live chat button doesn't do anything although it says it's online
- the Contact us page comes up with error Page Not Found
Hello,
Does anyone know if it's possible and how to enable the Message
Waiting Indicator (MWI) on a Sipura or Linksys ATA ?
I am looking in particular for the type of indicator that makes the
phone connected display something like New message on the screen, or
blink a light (depending on the
Hello,
I am trying to figure out a way to allow users listen to their
voicemails without having to enter a mailbox and a password when they
call from their own extension, while having to login with mailbox and
password when they call in from PSTN.
Any idea ? At least how to pass the mailbox and
]
Sent: Friday, November 09, 2007 4:39 PM
Subject: Re: [on-asterisk] Voicemail
On Nov 9, 2007 4:31 PM, Liviu Toma [EMAIL PROTECTED] wrote:
Any idea ? At least how to pass the mailbox and password to the
VoiceMail application automatically ?
Liviu,
You should be able to use the voicemailmain
Maybe the problem is with the line
exten = s,6,Flash()
Does the trunk you receive the incoming call from, support flash/call
transfer, etc ?
On Oct 30, 2007 11:36 AM, Anastasia LePlume [EMAIL PROTECTED] wrote:
Hello all!
My name Anastasia. Im new to asterisk and new to this group aswell. I am
From my experience with other telecom products, whenever a switch
needs to send DNIS information (coming through from a PRI for example)
to a device connected to an analog FXS port, the DNIS information gets
converted to plain DTMF tones sent in-band. Most if the time it's just
the plain digits.
Hi David,
I don't think Asterisk can work as a T.38 gateway yet. It can pass
through T.38 between 2 peers, but that's about it.
When it comes to T.38 gateways, Cisco is almost your only choice, they
hold probably 90% of the gateway market. Other companies/products that
can work as a T.38 gateway:
I am not too familiar with Cisco phones, but maybe it's something to
do with having call waiting enabled on each one of the lines ?
Liviu
On 9/26/07, Gary T. Giesen [EMAIL PROTECTED] wrote:
David,
Yes, I'm aware of that, but unfortunately it does two calls on each
line appearance (button),
If you choose to go with Philip's suggestion, certain phones (Nokia
for sure, even old ones) will allow you to send through a serial
cable, using AT commands (see below for the commands). I've done it
only in Windows, using hyper terminal, but I don't see why Linux
wouldn't work.
The problem is
The Telco will have to set it up that way (as a hunt group).
Liviu
On 8/9/07, Henry L.Coleman [EMAIL PROTECTED] wrote:
Another technical question
How do I provision a T1 to accept multiple channels originating from one DID
or put another way how do I offer a client a main number and two
My experience: old Pentium 4 @ 1.6 GHz (single core, one of the old
Socket 478), 1 GB RAM, Asterisk 1.4.9, Sangoma A101, during a Meetme
conference with 12 participants, the load average was showing: 0.07,
0.05, 0.00
Liviu
On 8/8/07, Henry L.Coleman [EMAIL PROTECTED] wrote:
Hi all, I have a
As far as I know Bell does SDSL at 2048 Kbps up/down, but only in the
downtown core. Not sure what the pricing is, but it may not fit the
$200 budget.
There's something interesting I learned a couple days ago: With some
ADSL providers in the GTA, you can do bonded DSL. It's not exactly
symetrical,
A friend of mine is looking for a SIP provider with low rates to India
(maybe a local company in India). Any suggestions ?
Thanks,
Liviu
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every second or so
== Primary D-Channel on span 1 up
== Primary D-Channel on span 1 up
== Primary D-Channel on span 1 up
However the D channel still shows down on the switch. I am guessing it't
going up and down very fast.
Thanks,
Liviu
On 7/4/07, Liviu Toma [EMAIL PROTECTED] wrote:
Hello
According to the Rogers website, the call display with name and number is
a different option (which costs $2 more) than the regular call display with
number only. Also, it's supported only on certain handsets:
Hello,
I have a problem with Asterisk 1.4 and a Sangoma T1 card (A101). Basically I
can't make the D channel come up no matter what I tried.
I installed the Zaptel driver, the Libpri and the wanpipe driver following
the instructions on Sangoma's web site:
Hello,
Does anyone have a sample provisioning file for the Grandstream GXP2000 ?
Thanks,
Liviu
Hi Christopher,
Vbuzzer uses port 80 for SIP messages, so under the peer details you have to
change change port=5060 to port=80. You probably have to do the same for user
details, I am not very good when it comes to TrixBox (I am using just a plain
asterisk), so I can't tell you for sure.
You
Hello,
I have a question more about telephony and long distance. A bit off
topic, I know, but I hope someone will help me with it.
When someone acquires a toll free number, they pay usually some long
distance carrier to provide it. However, I would assume that the call
from one end to another
I am using an Unlimitel account to call my Asterisk in order to be
called back and I am not being charged.
Regards,
Liviu Toma
On 5/30/07, Bruce Nik [EMAIL PROTECTED] wrote:
Just to add to Henry's post, when I said 1 minute of airtime will be used
was in regards to DID services from VoIP
You mean just route a single IP to a different network ?
From my knowledge about BGP, I don't think that would be possible.
Liviu
- Original Message -
From: Blaine Aldridge [EMAIL PROTECTED]
To: asterisk@uc.org; [EMAIL PROTECTED]
Sent: Monday, May 21, 2007 1:47 PM
Subject: Re:
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