Command line fu

[Hatem Nassrat]

This is my new favourite website:

    http://www.commandlinefu.com/commands/browse

Awesome sets of commands, my favourite must be:

    mplayer -vo caca some_file.avi

--
Hatem Nassrat
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[Richard Bonner]

On Fri, 11 Sep 2009, Hatem Nassrat wrote:

> This is my new favourite website:
>
>    http://www.commandlinefu.com/commands/browse
>
> Awesome sets of commands, my favourite must be:
>
>    mplayer -vo caca some_file.avi
>
> --
> Hatem Nassrat

***   The power is at the command line!

  Richard
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[_budman_]

Hatem Nassrat wrote:
> This is my new favourite website:
>
>     http://www.commandlinefu.com/commands/browse
>
> Awesome sets of commands, my favourite must be:
>
>     mplayer -vo caca some_file.avi
>
>  

Cool site. :)   Love using the cmd line.

Here's one I use at work.

Find broken symlinks  (not sure which version they added xtype, may not
work on all systems)
    find . -xtype l





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[Daniel Morrison-2]

2009/9/12 Rich <budman85@...>:
> Find broken symlinks  (not sure which version they added xtype, may not work on all systems)
>    find . -xtype l

-xtype has been in GNU findutils at least since the initial revision
was entered into their source code repository in 1996 (version 4.1).

But you might be on a Mac which lacks -xtype, so:

My reading of the man page suggests that -xtype may be a convenience option for:

find -L . -type l

...which is virtually the same as:

find . -follow -type l

Follow symbolic links, and check the type of the target. The target
can never be a link[*], so the type check always fails, _unless_ the
link could not be followed (i.e. is broken).

However I do find the man page to be quite complex and confusing, so I
may not have it completely correct. Seems to work for me.

-D.

[*] the target could be a link if you have an infinite loops of links
pointing back to each other, but in this case you get an error: "Too
many levels of symbolic links"
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[_budman_]

Daniel Morrison wrote:

2009/9/12 Rich budman85@...:
  

Find broken symlinks  (not sure which version they added xtype, may not work on all systems)
   find . -xtype l
    

-xtype has been in GNU findutils at least since the initial revision
was entered into their source code repository in 1996 (version 4.1).

  

Thanks, should of said other non-GNU versions. 

But you might be on a Mac which lacks -xtype, so:

My reading of the man page suggests that -xtype may be a convenience option for:

find -L . -type l

...which is virtually the same as:

find . -follow -type l

Follow symbolic links, and check the type of the target. The target
can never be a link[*], so the type check always fails, _unless_ the
link could not be followed (i.e. is broken).

However I do find the man page to be quite complex and confusing, so I
may not have it completely correct. Seems to work for me.
  

Thanks, I think I used the second suggestion before I found out how to use the xtype option.
I agree the documentation for that option needs clarification.






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[Hatem Nassrat]

On Fri, Sep 11, 2009 at 5:20 PM, Hatem Nassrat <hnassrat@...> wrote:
> This is my new favourite website:
>
>    http://www.commandlinefu.com/commands/browse

Vote for me:

    http://www.commandlinefu.com/commands/view/3467/send-echo-to-socket-network

--
Hatem Nassrat
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[Gerald-16]

I am looking for a SIP VoIP service I can use with a softphone on Linux
that has Nova Scotia numbers. What can people suggest?

Thanks

--
Gerald
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[Michael Gillie]

VoipGo.com

I use them myself. They are American, and VERY small. They have poor support, but somewhat reliable service.
I pay $17/mo for free long distance anywhere in NA.

 

Good luck,

 

Michael

 

On Sun, Sep 13, 2009 at 5:12 PM, Gerald <linux@...> wrote:

I am looking for a SIP VoIP service I can use with a softphone on Linux
that has Nova Scotia numbers. What can people suggest?

Thanks

--
Gerald
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--
Have a great day,


Michael C. Gillie

1-902-482-9644
Skype: hemmysoft

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[Scott Syms]

I've used voicenetwork.ca, but last time I checked they only had  
numbers for Halifax & Sydney.

tks

S

On 13-Sep-09, at 5:12 PM, Gerald wrote:

> I am looking for a SIP VoIP service I can use with a softphone on  
> Linux
> that has Nova Scotia numbers. What can people suggest?
>
> Thanks
>
> --
> Gerald
> _______________________________________________
> nSLUG mailing list
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[Daniel Morrison-2]

I'm using Les.net. Initial setup was a bit challenging (partly because
I'm using a cheapo VOIP adapter that they don't officially support).
But once working I've been very satisfied with the service over the
last year. $3.50/month and 1.5 cents/minute anywhere in NA. 1.1
cents/minute for local calls. Fractional minute billing. Free voice
mail.

I've heard of satisfied customers using link2voip.com.

Either way, Quality of Service on your local router is a must.

-D.

2009/9/13 Gerald <linux@...>:
> I am looking for a SIP VoIP service I can use with a softphone on Linux
> that has Nova Scotia numbers. What can people suggest?
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[Ian Campbell-9]

On Sun, Sep 13, 2009 at 05:29:47PM -0300, Daniel Morrison wrote:
> I'm using Les.net. Initial setup was a bit challenging (partly because
> I'm using a cheapo VOIP adapter that they don't officially support).
> But once working I've been very satisfied with the service over the
> last year. $3.50/month and 1.5 cents/minute anywhere in NA. 1.1
> cents/minute for local calls. Fractional minute billing. Free voice
> mail.

I used Les.net with a local asterisk setup and a Cisco 7961. It worked
fine, although voice quality was poor sometimes (tinny, symptom of
packet loss. Not bad enough to get choppy though.)

I think they were overloaded at the time, I would imagine they've
improved their capacity since then, it's been a couple years.

They have 902 numbers and if you call enough for it to make a
difference for a couple bucks a month you can get a flat rate.

Note that the flat rate lines allow for a lower number of concurrent
calls, if that matters to you.

> I've heard of satisfied customers using link2voip.com.

Yeah, I've heard of happy people on voip.ms too, no idea what their
number situation is.

> Either way, Quality of Service on your local router is a must.

This. A thousand times this.
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[Tim Johnson-13]

----- Original Message -----
From: "Gerald" <linux@...>
To: "Nova Scotia Linux User Group" <nslug@...>
Sent: Sunday, September 13, 2009 5:12 PM
Subject: [nSLUG] SIP VoIP service with NS phone number


> I am looking for a SIP VoIP service I can use with a softphone on Linux
> that has Nova Scotia numbers. What can people suggest?
>
> Thanks
>
> --
> Gerald

I use callwithus.com for my outgoing long distance. I put $10 in my account
back in January, still have lots left as there is no monthly fee. They also
sell DIDs but only have Halifax numbers if that makes any differance to you.

I also have a Washington state phone number, just because it was free:)  Go
to ipkall.com. They will want you to answer an incominjg call every once in
a while. I've let mine go for a bit over a month and it's still worked. I
use my work Vonage phone to call it on occation just to keep it active
(answer it or let it go to VM).  Does anyone else know of any free DID
providers, anywhere?

Tim



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[Daniel MacKay-2]

Maybe someone should start a VOIP society in Halifax, the members pay  
for a pri hooked to a voip / asterix box.  It would be good if someone  
could donate the hosting of course.
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[Gerald-16]

Thanks for the suggestions. I am trying out callcentric.com now and may
also evaluate voip.ms.
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[Hatem Nassrat]

On Mon, Sep 14, 2009 at 10:20 AM, Tim <tjllml@...> wrote:

>
> ----- Original Message -----
> From: "Gerald" <linux@...>
> To: "Nova Scotia Linux User Group" <nslug@...>
> Sent: Sunday, September 13, 2009 5:12 PM
> Subject: [nSLUG] SIP VoIP service with NS phone number
>
>
>> I am looking for a SIP VoIP service I can use with a softphone on Linux
>> that has Nova Scotia numbers. What can people suggest?
>>
>
> I use callwithus.com for my outgoing long distance. I put $10 in my account

Callwithus seems to be the only one out of the list of service
providers that is keeping with the times. They have a cool API, so you
can even easily write your own app that uses services such as:

    * Webcallback (aka. phone to phone)
    * usage & call history
    * rates ...

They also have all the information for integrating with asterisk and
many sip phones.

I am thinking of trying them out as soon as I run of outgoing credits
with voipstunt. I will be considering writing a QT app for the S60
that would utilise the webcallback feature, for those instances were
internet connectivity is limited.

--
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[Daniel Morrison-2]

2009/9/15 Hatem Nassrat <hnassrat@...>:

> Callwithus seems to be the only one out of the list of service
> providers that is keeping with the times. They have a cool API, so you
> can even easily write your own app that uses services such as:
>
>    * Webcallback (aka. phone to phone)
>    * usage & call history
>    * rates ...
>
> They also have all the information for integrating with asterisk and
> many sip phones.
>
> I am thinking of trying them out as soon as I run of outgoing credits
> with voipstunt. I will be considering writing a QT app for the S60
> that would utilise the webcallback feature, for those instances were
> internet connectivity is limited.

Can you explain what "webcallback" is? Try as I might, I can't imagine
what a VOIP service has to do with the WWW, and "phone to phone" also
sounds like, well, what a phone network does.

Perhaps I am not with the times, but I don't get what "writing your
own app" accomplishes. And what is an S60?

-D.
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[Hatem Nassrat]

Wow, reading your email, I realised my email must have seemed so cryptic :)

On Tue, Sep 15, 2009 at 11:10 PM, Daniel Morrison <draker@...> wrote:
> Can you explain what "webcallback" is? Try as I might, I can't imagine

Basically calls two numbers and joins them together.

[...]
> Perhaps I am not with the times, but I don't get what "writing your
> own app" accomplishes. And what is an S60?

Well what the app would accomplish is even when you have an
intermittent connection, usually on a mobile phone trying to connect
to a low signal rogue wireless access point (at-least it happens with
my phone) or just trying to save bandwidth costs. This app would
basically use the API to connect you to another party, using what they
call web callback.

S60 (Symbian 60) is an operating system for Nokia phones (and I
believe a few other phones as well).

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[Daniel Morrison-2]

2009/9/15 Hatem Nassrat <hnassrat@...>:
> On Tue, Sep 15, 2009 at 11:10 PM, Daniel Morrison <draker@...> wrote:
>> Can you explain what "webcallback" is? Try as I might, I can't imagine
>
> Basically calls two numbers and joins them together.

Thanks for the clarification. OK, so like an operator in the middle of a call.

> Well what the app would accomplish is even when you have an
> intermittent connection, usually on a mobile phone trying to connect
> to a low signal rogue wireless access point

Except I would call the client phone "rogue", rather than the access point!

> (at-least it happens with
> my phone) or just trying to save bandwidth costs.

SIP phone bandwidth must be pretty minimal... yeah, I guessed about
right. Say about 24 Kibps (kibi bits per second), or 3 KB/s. 7.5
GB/month if you use the phone continuously. (I used
http://www.asteriskguru.com/tools/bandwidth_calculator.php which I
found with google).

> This app would
> basically use the API to connect you to another party, using what they
> call web callback.

> S60 (Symbian 60) is an operating system for Nokia phones (and I
> believe a few other phones as well).

So... you are on an intermittent connection, possibly disappearing
from one IP network and appearing on another, and the client app on
your phone might interact with the third party in the middle
(webcallback), keeping (or "holding") your call active. So you don't
lose the call even when you drop out momentarily.

Am I close?

-D.
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[Hatem Nassrat]

On Tue, Sep 15, 2009 at 11:47 PM, Daniel Morrison <draker@...> wrote:
> So... you are on an intermittent connection, possibly disappearing
> from one IP network and appearing on another, and the client app on
> your phone might interact with the third party in the middle
> (webcallback), keeping (or "holding") your call active. So you don't
> lose the call even when you drop out momentarily.
>
> Am I close?

more or less, but you make it sound complicated :p

Basically once the phone-to-phone is initiated you no longer need a
network connection as the webservice would take over to call both
parties. To initiate the phone to phone with the API is like hitting a
single URL, so it should be quick and easy to do if you have any kind
of window while connected to an access point.

Some providers allow you to end the call on the server side, however,
that is not documented as an option with the CallWithUs API.

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[Daniel Morrison-2]

2009/9/15 Hatem Nassrat <hnassrat@...>:
> On Tue, Sep 15, 2009 at 11:47 PM, Daniel Morrison <draker@...> wrote:
>> So... you are on an intermittent connection, possibly disappearing
>> from one IP network and appearing on another, and the client app on
>> your phone might interact with the third party in the middle
>> (webcallback), keeping (or "holding") your call active. So you don't
>> lose the call even when you drop out momentarily.

> more or less, but you make it sound complicated :p

I make it sound complicated? Your way is magic:

> Basically once the phone-to-phone is initiated you no longer need a
> network connection

Quite incredible, to be able to communicate without a network connection!

> as the webservice would take over to call both
> parties. To initiate the phone to phone with the API is like hitting a
> single URL, so it should be quick and easy to do if you have any kind
> of window while connected to an access point.

Probably you mean something completely different from what I am
understanding, but I suspect you are living and breathing technology
and protocols that I have no clue about, so it all sounds impossible
to me... no matter. I don't really ever plan to roam with a VOIP
phone.

-D.

> Some providers allow you to end the call on the server side, however,
> that is not documented as an option with the CallWithUs API.
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