This is Kaleb, I have ABSOLUTELY no ties whatsoever with any VOIP service or
product, I am just an end-user. I currently use sixTel, I love it. I have
tried others and had very bad experiences, sellvoip.net being the absolute
worst from my experience. I haven't ever tried any of the
"unlimited"
services like broadvoice, I only have tried the pay-as-you-go providers that
support the IAX protocol. sixTel has been really good to me, they are a tad
slow on email support, but of you catch the through MSN messenger
(msn@sixtel.net) they usually are pretty good. I haven't dealt with their
service department for about over a month, I have had an account with them
for 2 months and had only one problem with inbound calls, their DIDs stopped
working for a small spell due to a problem with their carrier, beyond their
control (at least that is what they told me). When configuring their
service it was 100% required that in iax.conf the context was "sixTel"
with
the capital T or inbound wouldn't work, can't say that I have ever
noticed a
problem with outbound. I will send a copy of my sterilized sixTel config to
anyone that would like.
No Dan, I do not have a "tie" with them, I am just a happy customer.
I
especially like the fact that they email me automatically if they aren't
able to reach my server when someone calls. (has happened a time or two, was
MY fault). They also allow you to set up a fail-back number (can be pstn or
cellular or whatever) if your server is unreachable by them. They win my
business; If you don't like them, that is your call.
Kaleb
> I guess Kaleb has a tie up with them..
>
>
>
>I could never get my sixtel try call out except once. They customer service
>sucks as usual....
>
>
>
>I regret paying to them...Their sample config are all done from my side my
>my calls comes out with 'NO ANSWER"
>
>
>
>Dan
>
>
>
>On 25/01/06, Dovid Bender <asteriskdigium@yahoo.com> wrote:
>
>Let me guess you have no affiliation with them what so
>ever ? no commision on accounts either ?
>--- "Kaleb L. Kunzler"
><kunzlerklan.catchall@gmail.com> 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, I always get my calls, I couldn't think of
> a better provider. They
> are "picky" with the context that you use in your
> IAX.cc, but as long as you
> use the sample config that they provide it works
> beautifully. You only need
> $5 to open the account, that really isn't bad at
> all, others like
> sellvoip.net <http://sellvoip.net> (bad) require $25 to open an
account.
> If you need help getting
> their service to work for you, please contact me off
> list.
>
>
> >On 1/25/06, sdcharly@gmail.com <sdcharly@gmail.com>
> wrote:
> >
> > You can try with www,iax.cc too but i guesst not
> luck with a test
> > account..
> >
> > Dan
> >
> >
> > On 25/01/06, Nilesh Londhe <lvnilesh@gmail.com>
> wrote:
> > >
> > > I use www.voipjet.com and find it OK.
> > >
> > > On 1/24/06, Roberto Pereyra <
> pereyra.roberto@gmail.com> wrote:
> > > > Hi
>
> > > >
> > > > I looking a good IAX service for a emerging
> voip provider.
> > > >
> > > > Better with a test account to try.
> > > >
> > > > Thanks in advance.
> > > >
> > > > roberto
>
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Message: 10
Date: Thu, 26 Jan 2006 14:55:22 +0000
From: bails <bails@westcomuk.com>
Subject: Re: [Asterisk-Users] TDM400 pinout
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <43D8E2DA.4080603@westcomuk.com>
Content-Type: text/plain; charset=us-ascii; format=flowed
BJ Weschke wrote:> On 1/26/06, bails <bails@westcomuk.com> wrote:
>
>>Chris Bagnall wrote:
>>
>>>>Hi I'm looking for a pinout for the above. Note this has
>>>>what i'd call
>>>>RJ45 sockets (or someone smart can correct me). I need to
>>>>plug into BT (rj13?).
>>>
>>>
>>>Are you sure the TDM400 has RJ45 sockets? The pair I've got here
have
RJ12>>>sockets.
>>>
>>>I assume with the mention of BT, you're in the UK. The line is
on pins
2+5>>>of the BT connector, which'd usually translate to the 2 inner
pins of an
>>>RJ11 connector (pins 2+3). You should find an old modem cable will
do the
>>>job fine.
>>>
>>>If your TDM400 really does have RJ45 sockets, then you'd expect
the line
to>>>be on the middle pins (pins 4+5), similar to a modtap used in
structured
>>>cabling environments.
>>>
>>>Regards,
>>>
>>>Chris
>>
>>Thanks, yes they are rj45, we have had rj12 in he past I look at the
above.>>
>>Like I said though, pity Digium dont supply the information on there
>>site or with the cards, its a bit like everything in life today. We are
>>only the customer, but we're expected to do the running around.
>>
>
>
> Earlier versions of the TDM400 I believe were RJ45. They were changed
> to RJ11 I think I had heard at one point for compliance with some
> telco standards outside the US. But, in either case, yes, the middle
> pair is the "active" pair for your FXO/FXS ports on these cards
> whether RJ11 or RJ45.
>
> --
> Bird's The Word Technologies, Inc.
> http://www.btwtech.com/
> _______________________________________________
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>
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>
>
Thanks I can confirm that this is indeed correct
Bails
------------------------------
Message: 11
Date: Thu, 26 Jan 2006 07:56:16 -0700
From: Colin Anderson <ColinA@landmarkmasterbuilder.com>
Subject: RE: [Asterisk-Users] Bootable CD?
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users@lists.digium.com>
Message-ID:
<E251506D3758AA4882130317D52ACD3206DFA4@land-edm-hs2.landmarkhomes.net>
Content-Type: text/plain; charset="iso-8859-1"
To clarify: You have to write it as a DISK IMAGE. If you simply drag the ISO
file to your Nero project and write it, you will get a CD with a single file
on it - the ISO image - and not the CONTENTS of the ISO Image.
1. Run Nero
2. In the New Compilation dialog click Cancel
3. Click File > Burn Image, select" All Files" under Files of Type
and pick
your ISO.
4. Click OK, then click Write
hth
-----Original Message-----
From: sdcharly@gmail.com [mailto:sdcharly@gmail.com]
Sent: Thursday, January 26, 2006 1:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bootable CD?
yup... its a bootable image.. go ahead and just write it directly...
Dan
On 26/01/06, Sohail Arham < msarham@gmail.com
<mailto:msarham@gmail.com> >
wrote:
ahan...then it mean it doesnt need to uncompress it..juss write on cd by
nero burning software...??
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Message: 12
Date: Thu, 26 Jan 2006 09:58:07 -0500
From: Bill Michaelson <bill@cosi.com>
Subject: Re: [Asterisk-Users] * point to point t1 solution? /
alternatives
To: asterisk-users@lists.digium.com
Cc: vheether@ltps.org
Message-ID: <43D8E37F.3090605@cosi.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
This has been an interesting discussion for me (except for the
sniping). The last post led me, out of curiosity, to this wiki entry:
http://www.voip-info.org/wiki-Asterisk+TDMoE
I was unaware of this feature, and it looks pretty good. I've been
pondering replacing some T1's by leveraging IP capacity but of course
have run up against the QoS issue. My idea was different...
I don't have production experience with precisely this type of
application, but I ask for validation from this list. Pardon me for
stating what is undoubtedly obvious to many...
The key to assuring adequate performance in replacing a TDM link with IP
is to assure that adequate idle time is reserved for voice on the IP
segment(s) involved in the route. In this way, latency can be
stabilized, and if maintained below a certain (arbitrary) threshold,
performance can be deemed acceptable.
The first step, of course, is to assure that the virtual TDM allocation
does not exceed the available IP bandwidth (so leave a margin, which is
huge in the example given). The next step is to use routers which
respect the TOS field (however it is used; diffserv/whatever), and
finally, to assure that no non-VoIP traffic can be injected into the
path with higher routing priority.
On a point-to-point link, a pair of typical Linux boxes can do all
this. Given the original problem, I would place Asterisk boxes at
either end of the link, and have them blend the ordinary traffic with
the VoIP traffic (which would probably use IAX to relay calls between
the T1s), while assuring (enforcing) that VoIP packets are marked as
highest priority. There are varied ways of accomplishing this, and a
good reference which I've used in the past can be found at:
http://www.lartc.org/lartc.html
Additionally, I think one could use the tunneling techniques described
in that guide to encapsulate the non-VoIP traffic such that its packets'
originally marked TOS values are preserved for transit outside the
segment used for TDM emulation. In this way, that part of the segment
bandwidth not required for VoIP would function as a dedicated link,
allowing other prioritization of traffic such as interactive vs. bulk
(or even other voice!), with the added advantage that it could use the
reserved VoIP bandwidth when it is otherwise not required (albeit in the
case of a T-1 over 10Mb, that's insignificant).
Is this easier or harder than TDMoE as described? Does the TDMoE shared
idle bandwidth? What about stability (I'm thinking of SW releases)?
What other drawbacks or advantages are there?
>Date: Wed, 25 Jan 2006 23:53:59 -0700
>From: "Damon Estep" <damon@suburbanbroadband.net>
>Subject: [Asterisk-Users] * point to point t1 solution?
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
>
>Can anyone point me to a reference or sample config for bypassing a
>nailed up (point to point) t1 between two PBXs with asterisk and a pair
>of t1 cards?
>
>Right now I have 2 Nortel norstars connected to each other via a leased
>line t1. I also have a solid 10mbps low latency microwave link between
>the 2 sites.
>
>My goal is to run an asterisk box at each end with a t1 card and
>Ethernet card to act as a TDM<>SIP gateway to bypass the nailed T1 in
a
>relatively dumb configuration, with the goal of migrating off of the
>norstars eventually.
>
> In past situations I would have done this with a pair of Cisco routers
>with T1 interfaces in them but in this case I want to get asterisk into
>the picture as an eventual replacement for the norstars.
>
>
>
>
------------------------------
Message: 13
Date: Thu, 26 Jan 2006 10:08:44 -0500
From: Bill Michaelson <bill@cosi.com>
Subject: Re: [Asterisk-Users] * point to point t1 solution?
To: asterisk-users@lists.digium.com
Message-ID: <43D8E5FC.8040900@cosi.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
You've clarified your requirements for me. Please indulge me - I really
want to understand - what are the application implications of this? In
other words, what system behavioral changes will your users experience
in the various scenarios (pure circuit emulation vs. relay via IAX or
similar)?
Date: Thu, 26 Jan 2006 07:00:02 -0700
From: "Damon Estep" <damon@suburbanbroadband.net>
Subject: RE: [Asterisk-Users] * point to point t1 solution?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<07668904BA88BA4E9DA11CDE5B594CB20144E0BE@ns1.soho.soho-systems.com>
Content-Type: text/plain; charset="iso-8859-1"
Lets put the TDMoE aside for a minute...
The same trunking could be achieved with SIP or IAX, could it not (with
higher latency)?
The rest of the question remains - is there a way to get asterisk to output,
bit for bit, on a t1 interface, the same data that is input on a remote
asterisk box t1 interface - using any trunking protocol.
This is what would be required to truly emulate a "signaling un-aware"
point
to point t1 like one that you would get from a telco if you ordered a point
to point esf/b8zs t1 from A location to Z location.
Pure circuit emulation - not ISDN/CAS/E&M signaled voice.
Does that clarify the question at all?
------------------------------
Message: 14
Date: Thu, 26 Jan 2006 09:15:47 -0600
From: "Ross C" <wotech@cox.net>
Subject: RE: [Asterisk-Users] * point to point t1 solution?
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <000801c6228b$62f527b0$5f01a8c0@dtx1>
Content-Type: text/plain; charset="iso-8859-1"
Uhhhhhh..maybe you should ask Jean-Michel for a refund.
Wait, you havent paid a dime for this. Or Asterisk. Or most of the Asterisk
add-ons.
I always see people getting mad at other people for bad advice or bad
answers to their questions; people seem to forget that all this stuff is
FREE. If Jean-Michels advice isnt what youre looking for, say Thanks
for the info, but Id really like to know.. (geez, I feel like someones
mom). Hes taken time out of HIS day to try to help YOU for FREE.
If a high level of support and definitive answers are a must for your
situation, pay someone with experience, or see the following:
<expensive IP telephony>
http://www.cisco.com <http://www.cisco.com/>
http://www.nortel.com <http://www.nortel.com/>
http://www.inter-tel.com <http://www.inter-tel.com/>
http://www.avaya.com <http://www.avaya.com/>
http://www.3com.com <http://www.3com.com/>
</expensive IP telephony>
_____
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep
Sent: Thursday, January 26, 2006 3:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] * point to point t1 solution?
Actually, it is a quite appropriate response to ANYONE that includes this
type of comment in their reply
You probably need a couple of T1 cards, and some paid consulting to get it
working (I've never done it myself but that's how I would do it if I was
in
a hurry)
Perhaps something like this would have been better received;
I know it can (or cannot) be done, and here is the name of someone that
might be willing to help you for a fee
Look back though the archives and you will see that I have had some
participation here myself in the past
D
_____
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Simon Woodhead
Sent: Thursday, January 26, 2006 2:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * point to point t1 solution?
Bad day Damon? I think your comments are a little harsh towards someone who
is an active and informed contributor to the list. Jean-Michel could have
ignored you but he chose to share what he could. Maybe someone else will
have the complete answer to your question.
On 1/26/06, Damon Estep <damon@suburbanbroadband.net> wrote:
Jean-Michel,
You missed the entire point - the question is IS ASTERISK CAPABLE OF
EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB
BASED HINTS I MIGHT LOOK AT? Not WILL YOU DO IT FOR ME?
Your response to this post was un-informative and quite frankly it is the
type of useless response that most mailing lists and newsgroups could do
without.
Damon
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com <mailto:bounces@lists.digium.com> ] On
Behalf Of
Jean-Michel Hiver> Sent: Thursday, January 26, 2006 1:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] * point to point t1 solution?
>
> Damon Estep a icrit :
>
> > Can anyone point me to a reference or sample config for bypassing a
> > nailed up (point to point) t1 between two PBXs with asterisk and a
> > pair of t1 cards?
> >
> >
> >
> > Right now I have 2 Nortel norstars connected to each other via a
> > leased line t1. I also have a solid 10mbps low latency microwave link
> > between the 2 sites.
> >
> You probably need a couple of T1 cards, and some paid consulting to get
> it working (I've never done it myself but that's how I would do it
if I
> was in a hurry)
>
>
> > My goal is to run an asterisk box at each end with a t1 card and
> > Ethernet card to act as a TDM<>SIP gateway to bypass the nailed
T1 in
> > a relatively dumb configuration, with the goal of migrating off of the
> > norstars eventually.
> >
> If it's a point to point Asterisk <-> Asterisk configuration, why
use
> SIP rather than IAX? IAX configuration is very easy, so once you get the
> norstar <-> asterisk link up it'll be a piece of cake.
>
> Cheers,
> Jean-Michel.
>
> --
> Jean-Michel Hiver - http://ykoz.net/
> Dicouvrez la Riunion des Technologies IP & Telecom
> TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
>
>
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Message: 15
Date: Thu, 26 Jan 2006 15:16:25 -0000
From: "Steve Langstaff" <steve.langstaff@citel.com>
Subject: RE: [Asterisk-Users] * point to point t1 solution? /
alternatives
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<592CA2F7E2BFBD4088AC87ECFF34BECE1F3C52@not01-mxc01.citel.com>
Content-Type: text/plain; charset="iso-8859-1"
Remember, however that TDMoE is TDMoE, not TDMoIP - it's not routable
(unless you encapsulate it somehow, I guess).
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Bill
Michaelson
Sent: 26 January 2006 14:58
To: asterisk-users@lists.digium.com
Cc: vheether@ltps.org
Subject: Re: [Asterisk-Users] * point to point t1 solution? /
alternatives
This has been an interesting discussion for me (except for the
sniping). The last post led me, out of curiosity, to this wiki entry:
http://www.voip-info.org/wiki-Asterisk+TDMoE
I was unaware of this feature, and it looks pretty good. I've been
pondering replacing some T1's by leveraging IP capacity but of course
have run up against the QoS issue. My idea was different...
I don't have production experience with precisely this type of
application, but I ask for validation from this list. Pardon me for
stating what is undoubtedly obvious to many...
The key to assuring adequate performance in replacing a TDM link with IP
is to assure that adequate idle time is reserved for voice on the IP
segment(s) involved in the route. In this way, latency can be
stabilized, and if maintained below a certain (arbitrary) threshold,
performance can be deemed acceptable.
The first step, of course, is to assure that the virtual TDM allocation
does not exceed the available IP bandwidth (so leave a margin, which is
huge in the example given). The next step is to use routers which
respect the TOS field (however it is used; diffserv/whatever), and
finally, to assure that no non-VoIP traffic can be injected into the
path with higher routing priority.
On a point-to-point link, a pair of typical Linux boxes can do all
this. Given the original problem, I would place Asterisk boxes at
either end of the link, and have them blend the ordinary traffic with
the VoIP traffic (which would probably use IAX to relay calls between
the T1s), while assuring (enforcing) that VoIP packets are marked as
highest priority. There are varied ways of accomplishing this, and a
good reference which I've used in the past can be found at:
http://www.lartc.org/lartc.html
Additionally, I think one could use the tunneling techniques described
in that guide to encapsulate the non-VoIP traffic such that its packets'
originally marked TOS values are preserved for transit outside the
segment used for TDM emulation. In this way, that part of the segment
bandwidth not required for VoIP would function as a dedicated link,
allowing other prioritization of traffic such as interactive vs. bulk
(or even other voice!), with the added advantage that it could use the
reserved VoIP bandwidth when it is otherwise not required (albeit in the
case of a T-1 over 10Mb, that's insignificant).
Is this easier or harder than TDMoE as described? Does the TDMoE shared
idle bandwidth? What about stability (I'm thinking of SW releases)?
What other drawbacks or advantages are there?
>Date: Wed, 25 Jan 2006 23:53:59 -0700
>From: "Damon Estep" <damon@suburbanbroadband.net>
>Subject: [Asterisk-Users] * point to point t1 solution?
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
>
>Can anyone point me to a reference or sample config for bypassing a
>nailed up (point to point) t1 between two PBXs with asterisk and a pair
>of t1 cards?
>
>Right now I have 2 Nortel norstars connected to each other via a leased
>line t1. I also have a solid 10mbps low latency microwave link between
>the 2 sites.
>
>My goal is to run an asterisk box at each end with a t1 card and
>Ethernet card to act as a TDM<>SIP gateway to bypass the nailed T1 in
a
>relatively dumb configuration, with the goal of migrating off of the
>norstars eventually.
>
> In past situations I would have done this with a pair of Cisco routers
>with T1 interfaces in them but in this case I want to get asterisk into
>the picture as an eventual replacement for the norstars.
>
>
>
>
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***********************************************