Discussion:
iaxy vs sipura
(too old to reply)
Florin Andrei
2004-09-07 05:35:46 UTC
Permalink
I need a cheap simple adaptor for analog phones to use with Asterisk. It
should be some kind of "configure and forget" type of device, to use at
the office, or just throw it in a road warrior's bag and use it while
travelling, to call back to the "mothership".
I can't decide between iaxy and sipura. Can you guys help? Which one
would you use? (and why?)
I feel that iaxy might have an advantage while piercing through NAT
firewalls (at hotels and such), because of IAX, but i could be wrong.

Or can you recommend something else?
--
Florin Andrei

http://florin.myip.org/
Todd Lieberman
2004-09-07 10:42:10 UTC
Permalink
Post by Florin Andrei
I need a cheap simple adaptor for analog phones to use with Asterisk. It
should be some kind of "configure and forget" type of device, to use at
the office, or just throw it in a road warrior's bag and use it while
travelling, to call back to the "mothership".
I can't decide between iaxy and sipura. Can you guys help? Which one
would you use? (and why?)
I feel that iaxy might have an advantage while piercing through NAT
firewalls (at hotels and such), because of IAX, but i could be wrong.
Or can you recommend something else?
For configure and forget, I would not leave home w/out my IAXy.
Benjamin on Asterisk Mailing Lists
2004-09-07 12:54:08 UTC
Permalink
On Mon, 06 Sep 2004 22:35:46 -0700, Florin Andrei
Post by Florin Andrei
I need a cheap simple adaptor for analog phones to use with Asterisk. It
should be some kind of "configure and forget" type of device, to use at
the office, or just throw it in a road warrior's bag and use it while
travelling, to call back to the "mothership"
For travelling, no SIP based device will be "configure and forget".
Perhaps if you travel only within the US, you may be lucky most of the
time but pretty much anywhere else where IP addresses are scarce you
will be out of luck.

I have been travelling a lot on all inhabited continents, using hotel
provided internet connections, internet cafes, client's office LANs,
hotspots in public places, cafes, airports etc etc.

The most common experience is "SIP doesn't work at all" and the second
most common experience is "SIP only works after messing around a lot".
Even if you get SIP to work, you are likely to spend so much time on
fiddling with settings that it has a negative impact on your schedule.

I haven't used an IAXy yet but I run Asterisk on my Powerbook and use
IAX to connect back to my company's Asterisk server. That works all
the time and is "configure and forget". I assume it will be the same
when using an IAXy.

The only situation I can think of where the IAXy alone will not work
is with hotspots that require you to log in from a web browser in
order to activate the service, probably setting a cookie or something
like that. In this case you would need to run your IAXy on a NAT
provided by your notebook because the hotspot will not give you access
unless it sees the cookie or the MAC address of the machine that was
used to first sign in.

hope this helps
regards
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
Michael Bielicki
2004-09-07 12:57:44 UTC
Permalink
the only problem you will have whilst travelling with the iaxy is that
it supports only bandwidth hungry codecs. so if you are anywhere in the
world where bandwidth is a problem, the iaxy is a nogo
Post by Benjamin on Asterisk Mailing Lists
On Mon, 06 Sep 2004 22:35:46 -0700, Florin Andrei
Post by Florin Andrei
I need a cheap simple adaptor for analog phones to use with Asterisk. It
should be some kind of "configure and forget" type of device, to use at
the office, or just throw it in a road warrior's bag and use it while
travelling, to call back to the "mothership"
For travelling, no SIP based device will be "configure and forget".
Perhaps if you travel only within the US, you may be lucky most of the
time but pretty much anywhere else where IP addresses are scarce you
will be out of luck.
I have been travelling a lot on all inhabited continents, using hotel
provided internet connections, internet cafes, client's office LANs,
hotspots in public places, cafes, airports etc etc.
The most common experience is "SIP doesn't work at all" and the second
most common experience is "SIP only works after messing around a lot".
Even if you get SIP to work, you are likely to spend so much time on
fiddling with settings that it has a negative impact on your schedule.
I haven't used an IAXy yet but I run Asterisk on my Powerbook and use
IAX to connect back to my company's Asterisk server. That works all
the time and is "configure and forget". I assume it will be the same
when using an IAXy.
The only situation I can think of where the IAXy alone will not work
is with hotspots that require you to log in from a web browser in
order to activate the service, probably setting a cookie or something
like that. In this case you would need to run your IAXy on a NAT
provided by your notebook because the hotspot will not give you access
unless it sees the cookie or the MAC address of the machine that was
used to first sign in.
hope this helps
regards
benjk
Florin Andrei
2004-09-07 17:31:58 UTC
Permalink
Post by Michael Bielicki
the only problem you will have whilst travelling with the iaxy is that
it supports only bandwidth hungry codecs. so if you are anywhere in the
world where bandwidth is a problem, the iaxy is a nogo
Would iaxy work over a plain dialup connection? 56k? 33k?
(assuming the bandwidth is fine between the ISP and the location of the
Asterisk server)
--
Florin Andrei

http://florin.myip.org/
Eric Wieling
2004-09-07 17:43:43 UTC
Permalink
Post by Florin Andrei
Would iaxy work over a plain dialup connection? 56k? 33k?
(assuming the bandwidth is fine between the ISP and the location of the
Asterisk server)
No. The IAXy supports only G711 and ADPCM or G726 (I don't recall
which). Both take MUCH more bandwidth than dialup can provide.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
Benjamin on Asterisk Mailing Lists
2004-09-07 17:51:02 UTC
Permalink
On Tue, 07 Sep 2004 10:31:58 -0700, Florin Andrei
Post by Florin Andrei
Would iaxy work over a plain dialup connection? 56k? 33k?
(assuming the bandwidth is fine between the ISP and the location of the
Asterisk server)
The IAXy supports ADPCM which is 32k. Considering the IP overhead you
wouldn't get it to work over a 33k dialup connection, but it should
work just fine on 56k.

rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
Lyle Giese
2004-09-07 20:40:00 UTC
Permalink
Hmmm, I thought 56k modems were 56k outbound only and maxs at 33k inbound
or did the standard change again when I wasn't looking?

And besides when did you get better than 28.8 through a hotel PBX? >33k
with a 56k modem in the real world is not that common.

Lyle

----- Original Message -----
From: "Benjamin on Asterisk Mailing Lists" <***@gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-***@lists.digium.com>
Sent: Tuesday, September 07, 2004 12:51 PM
Subject: Re: [Asterisk-Users] iaxy vs sipura
Post by Benjamin on Asterisk Mailing Lists
On Tue, 07 Sep 2004 10:31:58 -0700, Florin Andrei
Post by Florin Andrei
Would iaxy work over a plain dialup connection? 56k? 33k?
(assuming the bandwidth is fine between the ISP and the location of the
Asterisk server)
The IAXy supports ADPCM which is 32k. Considering the IP overhead you
wouldn't get it to work over a 33k dialup connection, but it should
work just fine on 56k.
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
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Scott Lykens
2004-09-07 20:44:12 UTC
Permalink
Post by Lyle Giese
Hmmm, I thought 56k modems were 56k outbound only and maxs at 33k inbound
or did the standard change again when I wasn't looking?
And besides when did you get better than 28.8 through a hotel PBX? >33k
with a 56k modem in the real world is not that common.
When I first read the above posts I thought as you did but found
through some Googling that v.92 is 56k (53k) downstream and up to 48k
upstream.

Also, a note that the dialup connection will add in the neighborhood
of 140-170 ms of latency. If this is on top of an already high delay
connection you might find the conversation to be more of a
walkie-talkie type rather than natural flow, assuming you are using a
codec that will function over dialup to begin with.
Benjamin on Asterisk Mailing Lists
2004-09-08 03:24:15 UTC
Permalink
Post by Lyle Giese
Hmmm, I thought 56k modems were 56k outbound only and maxs at 33k inbound
or did the standard change again when I wasn't looking?
And besides when did you get better than 28.8 through a hotel PBX? >33k
with a 56k modem in the real world is not that common.
Don't mix up the content of different posts.

In one post, I explained how I used IAX and ILBC successfully on a sub
20k dialup link in Egypt (that was not a hotel but an office, BTW) and
in a different paragraph I mentioned hotels in foreign countries as
one of many locations where things can go funny.

In another post, I responded to a question whether or not it might be
possible to use an IAXy on a 33k or 56k dialup connection. There was
no mentioning of any hotels and I did say that I have not used an IAXy
myself. My comment that the IAXy *should* work on a 56k link was based
on the fact that it supports ADPCM and if you check out the specs of
most DSPs which do ADPCM you will find that they do 16, 24, 32 and 40
kbps, although the ITU ADPCM recommendation (G.726) talks about 32
kbps. Which chip the IAXy uses and which modes that chip supports, I
guess only Digium can tell you. If you want to be absolutely sure,
then you may just want to try it out.

In fact, I think it would be very welcome if somebody with actual
experience testing the IAXy on the road posted some feedback here.
There is nothing that beats testing in the field.

rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
Brian Capouch
2004-09-07 13:14:57 UTC
Permalink
Post by Benjamin on Asterisk Mailing Lists
On Mon, 06 Sep 2004 22:35:46 -0700, Florin Andrei
Post by Florin Andrei
I need a cheap simple adaptor for analog phones to use with Asterisk. It
should be some kind of "configure and forget" type of device, to use at
the office, or just throw it in a road warrior's bag and use it while
travelling, to call back to the "mothership"
For travelling, no SIP based device will be "configure and forget".
Perhaps if you travel only within the US, you may be lucky most of the
time but pretty much anywhere else where IP addresses are scarce you
will be out of luck.
If you have a Linux laptop with you, then in fact the SIP devices can be
configured to "hide" behind it. The laptop can then run an instance of
asterisk that connects to the "home" asterisk server, and with the
asterisk server on the laptop handling the NAT issues, it's pretty much
plug and play anywhere in the world. I've even had mine going when I'm
in a "double-NAT" situation.

This overcomes the codec situation with the iaxy mentioned in another
mail in this thread.

B.
Benjamin on Asterisk Mailing Lists
2004-09-07 14:57:26 UTC
Permalink
Post by Brian Capouch
If you have a Linux laptop with you, then in fact the SIP devices can be
configured to "hide" behind it. The laptop can then run an instance of
asterisk that connects to the "home" asterisk server,
Like I said: I run Asterisk on my Powerbook to do IAX to my company's
Asterisk server.

Keep in mind though that you don't need to have a Linux notebook to do
this. A Powerbook running MacOSX runs Asterisk just fine. This may not
be much of an issue for the Linux geeks and techies on the list, but
if you have to send sales people and other non-tech folks on business
trips and give them something to connect, then probably a Powerbook
running OSX will be an easier choice since they get to keep their
native MS-Office.

If there is sufficient interest, I'll be happy to write an IAX Peering
Assistant for OSX so that non-tech folks can set this up by
themselves. Anybody interested may drop me an email if they wish.
Post by Brian Capouch
and with the
asterisk server on the laptop handling the NAT issues, it's pretty much
plug and play anywhere in the world. I've even had mine going when I'm
in a "double-NAT" situation.
I have tested IAX with six NAT levels (before we ran out of NAT
routers). NAT is no problem for IAX, regardless how many levels. The
only challenge are services that require you to sign in using a web
browser to set cookies or record MAC addresses of the machine signed
in from.

This would pose a problem for the IAXy because it hasn't got a browser
(not that it should). This problem is of course solved with the above
mentioned notebook gateway solution, too.
Post by Brian Capouch
This overcomes the codec situation with the iaxy mentioned in another
mail in this thread.
I'd think that this is not a real problem because those places where
bandwidth is a problem are likely the places where you would need to
bring your own computer anyway.

As for challenging situations ...

I have used IAX and ILBC between Cairo, Egypt and Tokyo on what has
got to be one of the world's worst dialup connections. I am talking
about 3-4 kbps before and 15-18 kbps after replacing the phone wiring
in the building. It was so bad that SMTP always timed out and HTTP
required you to hit reload four or five times before you got anything
in your browser. Yet, IAX/ILBC worked like a Swiss clock. I had many
hour long phone calls and the voice quality was almost
indistinguishable from first world PSTNs. The only thing that made us
notice there was a problem with the connection was occassional lag.
SIP never worked once in Egypt, no matter what we tried.

SIP is another one of those overengineered things which only work in
the most ideal first world situations but become highly unreliable
when used in the rest of the world. IAX just works everywhere all the
time every time.

rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
Andy Powell
2004-09-10 12:05:09 UTC
Permalink
Post by Benjamin on Asterisk Mailing Lists
Post by Brian Capouch
If you have a Linux laptop with you, then in fact the SIP devices can be
configured to "hide" behind it. The laptop can then run an instance of
asterisk that connects to the "home" asterisk server,
Like I said: I run Asterisk on my Powerbook to do IAX to my company's
Asterisk server.
Keep in mind though that you don't need to have a Linux notebook to do
this. A Powerbook running MacOSX runs Asterisk just fine. This may not
be much of an issue for the Linux geeks and techies on the list, but
if you have to send sales people and other non-tech folks on business
trips and give them something to connect, then probably a Powerbook
running OSX will be an easier choice since they get to keep their
native MS-Office.
At the risk of stating the obvious.... if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever
and errm... problem solved.. :D

Andy
Benjamin on Asterisk Mailing Lists
2004-09-10 13:02:55 UTC
Permalink
On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell
Post by Andy Powell
At the risk of stating the obvious.... if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever
and errm... problem solved.. :D
Certainly an option, but most business folks will want to have their
Outlook contacts and Excel spreadsheets in front of them when they are
on the phone. Dual boot environments are not ideal in those
situations. Imagine you're talking to some guy on the phone about
prices and he tells you "I cant' tell you what the discounts are right
now because I would have to shut down the phone system to open Excel".

However, you could use VMware on an Intel notebook to run both Windoze
and Linux concurrently. This wouldn't be ideal for a real PBX for
performance reasons, but since all you are going to use Asterisk for
is to be a gateway for one single user, it's probably ok in this
particular scenario.

I remember there was a guy in Romania who reported he had VMware with
Windoze and Asterisk on Linux running as a home PBX on his PC and it
seemed to be alright.

If you'd combine such a setup with a Windoze GUI tool that will start
and stop the Linux environment and Asterisk at the push of a button,
then you'd have a fairly convenient and workable SIP/IAX gateway
solution for travelling biz folks.

rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
Leo Ann Boon
2004-09-08 00:06:18 UTC
Permalink
Post by Benjamin on Asterisk Mailing Lists
I have been travelling a lot on all inhabited continents, using hotel
provided internet connections, internet cafes, client's office LANs,
hotspots in public places, cafes, airports etc etc.
The most common experience is "SIP doesn't work at all" and the second
most common experience is "SIP only works after messing around a lot".
Even if you get SIP to work, you are likely to spend so much time on
fiddling with settings that it has a negative impact on your schedule.
Not so true. If you can use a SIP ALG (or far end NAT traversal device
like Jasomi), it will pretty much make the config problems 'disappear'.
I've had very good results using SER with nathelper module. My setup-
SER handles the NAT traversal and external extensions while * handles
the office extensions, voicemail and PSTN gw. So far, I have yet to
encounter a scenario that doesn't work with this setup. Most of the
time, my colleagues can walk into a customer's premise, turn on the
Cisco ATA and viola - they can call the office or make PSTN calls.
Benjamin on Asterisk Mailing Lists
2004-09-08 05:34:32 UTC
Permalink
Post by Leo Ann Boon
Post by Benjamin on Asterisk Mailing Lists
I have been travelling a lot on all inhabited continents, using hotel
provided internet connections, internet cafes, client's office LANs,
hotspots in public places, cafes, airports etc etc.
The most common experience is "SIP doesn't work at all" and the second
most common experience is "SIP only works after messing around a lot".
Even if you get SIP to work, you are likely to spend so much time on
fiddling with settings that it has a negative impact on your schedule.
Not so true. If you can use a SIP ALG (or far end NAT traversal device
like Jasomi), it will pretty much make the config problems 'disappear'.
If I had the money to afford Jasomi gear, I wouldn't care about the
phone bill I clock up in some hotel. In fact, I could spend hours
talking on the outrageously overpriced GSM roaming service and it
would still be cheaper than buying Jasomi.

Besides, NAT is not the only trouble you will have with SIP when you
travel to non-OECD places. Jasomi does nothing to get you past those
other hurdles.
Post by Leo Ann Boon
I've had very good results using SER with nathelper module. My setup-
SER handles the NAT traversal and external extensions while * handles
the office extensions, voicemail and PSTN gw. So far, I have yet to
encounter a scenario that doesn't work with this setup. Most of the
time, my colleagues can walk into a customer's premise, turn on the
Cisco ATA and viola - they can call the office or make PSTN calls.
All depends on where they are going. I could send them on a month long
trip where they never stay in the same place twice and they wouldn't
get it working once.

I have spent hundreds of hours in third world environments trying to
get VoIP connections going from ad hoc offices, public places and
customer premises. I am not talking from the convenience of my first
world infrastructure home or office theorising about why things should
work overseas in less fortunate places. I have actually been to those
places and the experience is precisely like I said, no buts and ifs
whatsoever.

Let me give you some examples ...

I have been to hotels which had broadband in the room and I paid so
much extra for that broadband room that it really didn't make sense
because I wouldn't have been able to make up for the extra cost by
saving on phone calls. However, I paid for it because I wanted to try
it out.

In many of those places, you will find yourself behind two levels of
NAT. It is also not uncommon to find that they block ports you need or
they reset their routers periodically every 60 seconds.

For example, many hotels in Hong Kong use a provider specialising in
serving hotels and they reset their routers every 60 seconds probably
to prevent customers from watching video streams or making VoIP calls
because the hotels want to make money on their inhouse services.

As you would expect, any services orther than HTTP will be severely
impeded. SSH sessions will freeze, tunneling is out of the question
and ftp downloads are problematic.

Yet IAX connections survived this ordeal usually 15-20 times before
the connection terminated. SIP connections never did. In other words,
your SIP call will last exactly 1 minute while you can talk about 15
to 20 minutes on an IAX call.

And you ain't seen nothing yet if you haven't been to the Middle East
and Africa.

I was in Saudi Arabia for one of the main ISPs evaluating their
options for running a domestic-only VoIP service, so I am not talking
about sitting in some hotel trying to make things work on a badly run
dialup service.

All ISPs in the KSA go through the national internet backbone which is
run by the government and like all things there it is heavily
restricted. Traditional VoIP won't work unless you get the government
backbone to co-operate. For that to happen, you'd probably have to get
an invitation to install a system for the royal family. Yet, there is
a rather simple trick how you can get a connection going using IAX. I
won't reveal this here though for obvious reasons. All I can say is
the same trick will not work for SIP. Oh, BTW, tunneling is a no-go
there, too.

Another interesting environment is Egypt. Things are not restricted
there as they are in the KSA, but the infrastructure is mostly
unsuitable to do VoIP. ISPs in Egypt are very inventive to overcome
the lack of resources they face. You will find this kind of
inventiveness in many third world environments. Unfortunately, the
things they do will pose challenges that SIP simply cannot cope with
in a reliable fashion.

The observations you make on Egyptian internet connections are so
weird, so plenty and so random that it is impossible to present even
only an outline within the scope of a posting like this. So, I will
keep this short and just give you a minor non-representative but easy
to describe example:

I was doing some troubleshooting of an IP connection between Cairo and
Alexandria, the two major cities in Egypt, only about 200 km apart.
The first surprise was that traceroute showed the connection was going
first to the US via the UK only to come back to the UK and then back
to Egypt via Turkey. The next surprise was that no consecutive
traceroute would show the same route. Every time the route was
different, often jumping in and out of third countries. The latency
was going up and down like a rollercoaster. Depening on the route, RTP
would pass through or not and what have you not.

In other words, when you see those things happending in front of your
eyes you are just about to give up and say "There is NO WAY I am going
to get any VoIP connection going over THIS". And, indeed, the
connection barely delivers anything other than ping and traceroute.

I am always as surprised as anybody else to see that IAX can cope with
such challenges. IAX never fails to amaze me. Everytime I run into
another outrageous situation, I think "This is it. This time it won't
work. No chance." and everytime IAX proves me wrong.

These are just a few examples out of many places I have been to in
this region and other regions. They are representative in the sense
that the real world out there is far more diverse and throws problems
at you that no theorising from the comfort of an OECD country with
excellent infrastructure could possibly foresee. You have to acutally
go out there and visit these places to see for yourself that SIP is
not a universally suitable instrument.

You may say this is all just because I don't know how to make SIP work
and I will give you this: I have come to not bothering with SIP
anymore if I can use IAX and that means I spend much less time on
troubleshooting SIP these days. But even if we assume that I don't
know anything about SIP, then it is still going to be a testament for
IAX' reliability, robustness and user friendliness because I didn't
know anything about IAX at the time I ran into these problems either.
Yet I was able to get IAX working where SIP didn't work and I spent an
order of magnitude more time on troubleshooting SIP than I did on
setting up IAX in its place after giving up.

Keep in mind though that I have been working with engineers on the
ground who know a lot more about SIP than I do and quite a few of them
know at least as much as the resident SIP gurus on this list. Often
they are trained by the big names in the VoIP industry or they are
ex-Cisco engineers etc. If they can't make it work with SIP and I can
with IAX, again it is a testament to IAX' reliability, robustness and
user friendliness.

Besides, we were talking about "configure and forget". All I said was
that if you venture outside of the US, SIP will not be "configure and
forget" where IAX will be.

rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
Rich Adamson
2004-09-08 01:13:34 UTC
Permalink
Post by Benjamin on Asterisk Mailing Lists
Post by Florin Andrei
I need a cheap simple adaptor for analog phones to use with Asterisk. It
should be some kind of "configure and forget" type of device, to use at
the office, or just throw it in a road warrior's bag and use it while
travelling, to call back to the "mothership"
For travelling, no SIP based device will be "configure and forget".
Perhaps if you travel only within the US, you may be lucky most of the
time but pretty much anywhere else where IP addresses are scarce you
will be out of luck.
Believe or not, I've been using a snom 200 configured with nat and haven't found
any US locations where it doesn't work with a couple of exceptions.

One exception is some hotels that use a web redirect service to basically
authenticate a session "before" the user is allowed to do anything. Examples
include some Marriotts, very small number of Holiday Inn Express's, etc. In
those cases, at least "one" web attempt must be made before a clear channel
is allowed from the dhcp assigned address.

The snom is never reconfigured (other then some corporations/institutions don't
support dhcp), and just seems to work with an * server on a registered IP.

I'm writing this from a hotel that only has wireless, so it doesn't work here.
But, xlite sort-of-kind-of works in its place. :)

Rich
Benjamin on Asterisk Mailing Lists
2004-09-08 03:02:34 UTC
Permalink
Post by Rich Adamson
Believe or not, I've been using a snom 200 configured with nat and haven't found
any US locations where it doesn't work with a couple of exceptions.
Read my post again and you will find that I was talking about
locations *outside* of the US.

rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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Brian West
2004-09-08 03:36:39 UTC
Permalink
ADPCM != G.726 they are a little bit different. codec_adpcm.c and
codec_g726.c

bkw
-----Original Message-----
Sent: Tuesday, September 07, 2004 10:24 PM
To: Lyle Giese
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iaxy vs sipura
Post by Lyle Giese
Hmmm, I thought 56k modems were 56k outbound only and maxs at 33k
inbound
Post by Lyle Giese
or did the standard change again when I wasn't looking?
And besides when did you get better than 28.8 through a hotel PBX? >33k
with a 56k modem in the real world is not that common.
Don't mix up the content of different posts.
In one post, I explained how I used IAX and ILBC successfully on a sub
20k dialup link in Egypt (that was not a hotel but an office, BTW) and
in a different paragraph I mentioned hotels in foreign countries as
one of many locations where things can go funny.
In another post, I responded to a question whether or not it might be
possible to use an IAXy on a 33k or 56k dialup connection. There was
no mentioning of any hotels and I did say that I have not used an IAXy
myself. My comment that the IAXy *should* work on a 56k link was based
on the fact that it supports ADPCM and if you check out the specs of
most DSPs which do ADPCM you will find that they do 16, 24, 32 and 40
kbps, although the ITU ADPCM recommendation (G.726) talks about 32
kbps. Which chip the IAXy uses and which modes that chip supports, I
guess only Digium can tell you. If you want to be absolutely sure,
then you may just want to try it out.
In fact, I think it would be very welcome if somebody with actual
experience testing the IAXy on the road posted some feedback here.
There is nothing that beats testing in the field.
rgds
benjk
--
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Tokyo, Japan.
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John Kington
2004-09-09 16:52:56 UTC
Permalink
At 09:54 PM 9/7/2004 +0900, Benjamin on Asterisk Mailing Lists
Post by Benjamin on Asterisk Mailing Lists
For travelling, no SIP based device will be "configure and forget".
Perhaps if you travel only within the US, you may be lucky most of the
time but pretty much anywhere else where IP addresses are scarce you
will be out of luck.
I have been travelling a lot on all inhabited continents, using hotel
provided internet connections, internet cafes, client's office LANs,
hotspots in public places, cafes, airports etc etc.
The most common experience is "SIP doesn't work at all" and the second
most common experience is "SIP only works after messing around a lot".
What about sip softphones that use STUN? I am especially interested in UK
because
my daughter is going to study in London.

Regards,
John
Benjamin on Asterisk Mailing Lists
2004-09-09 17:23:08 UTC
Permalink
Post by John Kington
What about sip softphones that use STUN? I am especially interested in UK
because my daughter is going to study in London.
If she is going to be on a residential ADSL, that shouldn't be a
problem. I have friends in the UK who use both softphones and
Grandstreams behind NAT on BT's ADSL service and we haven't had any
major problems other than softphones locking up the PC or some silly
stuff like that.

Just make sure your Asterisk server is on a public IP address, or if
it is behind NAT, then let her use FWD and use IAX to connect your
Asterisk server to FWD.

rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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John Kington
2004-09-10 21:54:37 UTC
Permalink
Post by Benjamin on Asterisk Mailing Lists
Post by John Kington
What about sip softphones that use STUN? I am especially interested in UK
because my daughter is going to study in London.
If she is going to be on a residential ADSL, that shouldn't be a
problem. I have friends in the UK who use both softphones and
Grandstreams behind NAT on BT's ADSL service and we haven't had any
major problems other than softphones locking up the PC or some silly
stuff like that.
Just make sure your Asterisk server is on a public IP address, or if
it is behind NAT, then let her use FWD and use IAX to connect your
Asterisk server to FWD.
I have asterisk running on a machine with a public ip which is pointed to by
dyndns. My wife's sister in France can call using sjphone (sip soft phone).
I have had no problems using asterisk for these calls nor for long distance
through Gafachi. (Did I just be bannished from the mailing list?)
My daughter has a XP laptop with 802.11g. I don't know what Internet access
she will have if any in the student housing. I am hoping she can find a
wireless
connection that she can share. I would like to also make it so she can walk
into some place that is a hotspot and make VoIP calls. Would a soft phone
using IAX be more reliable (easier) to use in this situation? I am sure she
will encounter NAT at least once in either situation. Her room does have a
port for telephone but I think she is on her own to get any phone service.
I have signed up with FWD and also have a DID through CallUK. She can
always use that to call back to the US.
Regards,
John
Bill Seddon
2004-09-10 13:34:00 UTC
Permalink
I run Asterisk on Redhat 8.0 with a VM hosted by Microsoft's Virtual PC
which, in turn, runs on Windows 2000 Server. Works like a charm. Can't use
Zaptel cards but that's OK for me. I can put it into standby any time and
it takes only a few seconds to start up the VM from its saved state and at
that time the Linux session (and Asterisk) is available once again.

Bill Seddon

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Benjamin on
Asterisk Mailing Lists
Sent: September 10, 2004 2:03 PM
To: Andy Powell
Cc: asterisk-***@lists.digium.com
Subject: Re: [Asterisk-Users] iaxy vs sipura

On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell
Post by Andy Powell
At the risk of stating the obvious.... if you have a laptop not running
MacOSX (ie perhaps running windows) download my asterisk live! cd (
http://www.automated.it/asterisk/ ), burn it and test it on your laptop and
bung it in your laptop case along with your iaxy/sipura/whatever
Post by Andy Powell
and errm... problem solved.. :D
Certainly an option, but most business folks will want to have their
Outlook contacts and Excel spreadsheets in front of them when they are
on the phone. Dual boot environments are not ideal in those
situations. Imagine you're talking to some guy on the phone about
prices and he tells you "I cant' tell you what the discounts are right
now because I would have to shut down the phone system to open Excel".

However, you could use VMware on an Intel notebook to run both Windoze
and Linux concurrently. This wouldn't be ideal for a real PBX for
performance reasons, but since all you are going to use Asterisk for
is to be a gateway for one single user, it's probably ok in this
particular scenario.

I remember there was a guy in Romania who reported he had VMware with
Windoze and Asterisk on Linux running as a home PBX on his PC and it
seemed to be alright.

If you'd combine such a setup with a Windoze GUI tool that will start
and stop the Linux environment and Asterisk at the push of a button,
then you'd have a fairly convenient and workable SIP/IAX gateway
solution for travelling biz folks.

rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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hank smith
2004-09-10 17:07:09 UTC
Permalink
how much ram you got on the pc running the vm? also will microsoft Virtual
PC run on xp home?
thanks
hank
----- Original Message -----
From: "Bill Seddon" <***@lyquidity.com>
To: "'Benjamin on Asterisk Mailing Lists'" <***@gmail.com>;
"'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-***@lists.digium.com>
Sent: Friday, September 10, 2004 6:34 AM
Subject: RE: [Asterisk-Users] iaxy vs sipura
Post by Bill Seddon
I run Asterisk on Redhat 8.0 with a VM hosted by Microsoft's Virtual PC
which, in turn, runs on Windows 2000 Server. Works like a charm. Can't use
Zaptel cards but that's OK for me. I can put it into standby any time and
it takes only a few seconds to start up the VM from its saved state and at
that time the Linux session (and Asterisk) is available once again.
Bill Seddon
-----Original Message-----
Asterisk Mailing Lists
Sent: September 10, 2004 2:03 PM
To: Andy Powell
Subject: Re: [Asterisk-Users] iaxy vs sipura
On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell
Post by Andy Powell
At the risk of stating the obvious.... if you have a laptop not running
MacOSX (ie perhaps running windows) download my asterisk live! cd (
http://www.automated.it/asterisk/ ), burn it and test it on your laptop and
bung it in your laptop case along with your iaxy/sipura/whatever
Post by Andy Powell
and errm... problem solved.. :D
Certainly an option, but most business folks will want to have their
Outlook contacts and Excel spreadsheets in front of them when they are
on the phone. Dual boot environments are not ideal in those
situations. Imagine you're talking to some guy on the phone about
prices and he tells you "I cant' tell you what the discounts are right
now because I would have to shut down the phone system to open Excel".
However, you could use VMware on an Intel notebook to run both Windoze
and Linux concurrently. This wouldn't be ideal for a real PBX for
performance reasons, but since all you are going to use Asterisk for
is to be a gateway for one single user, it's probably ok in this
particular scenario.
I remember there was a guy in Romania who reported he had VMware with
Windoze and Asterisk on Linux running as a home PBX on his PC and it
seemed to be alright.
If you'd combine such a setup with a Windoze GUI tool that will start
and stop the Linux environment and Asterisk at the push of a button,
then you'd have a fairly convenient and workable SIP/IAX gateway
solution for travelling biz folks.
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users
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Greg Boehnlein
2004-09-11 04:13:28 UTC
Permalink
Post by Benjamin on Asterisk Mailing Lists
However, you could use VMware on an Intel notebook to run both Windoze
and Linux concurrently. This wouldn't be ideal for a real PBX for
performance reasons, but since all you are going to use Asterisk for
is to be a gateway for one single user, it's probably ok in this
particular scenario.
Or you could use AstWind, which runs concurrently with Windows and is
built entirely on Open Source software (CoLinux Kernel, Debian, Asterisk)
and avoid paying for Vmware! ;)

Plus, installation is a snap.

See Digium's press release:
http://www.digium.com/index.php?menu=astwind

You can find more information on AstWind at:
http://www.voip-info.org/wiki-AstWind
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
KP-216-121-ST
hank smith
2004-09-11 05:14:03 UTC
Permalink
I have to have access to sound on linux to use the screen reader for linux
and from what I under stand colinux don't support sound.
otherwise this would be the perfict sullution.
----- Original Message -----
From: "Greg Boehnlein" <***@nacs.net>
To: "Benjamin on Asterisk Mailing Lists" <***@gmail.com>;
"Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-***@lists.digium.com>
Cc: "Andy Powell" <***@beagles-den.demon.co.uk>
Sent: Friday, September 10, 2004 9:13 PM
Subject: Re: [Asterisk-Users] iaxy vs sipura
Post by Greg Boehnlein
Post by Benjamin on Asterisk Mailing Lists
However, you could use VMware on an Intel notebook to run both Windoze
and Linux concurrently. This wouldn't be ideal for a real PBX for
performance reasons, but since all you are going to use Asterisk for
is to be a gateway for one single user, it's probably ok in this
particular scenario.
Or you could use AstWind, which runs concurrently with Windows and is
built entirely on Open Source software (CoLinux Kernel, Debian, Asterisk)
and avoid paying for Vmware! ;)
Plus, installation is a snap.
http://www.digium.com/index.php?menu=astwind
http://www.voip-info.org/wiki-AstWind
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
http://www.n2net.net Where everything clicks into place!
KP-216-121-ST
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