[sip-comm] Announcing translate.sip-communicator.org and call for contributors


#1

Hello,

I'm happy to announce that we've setup an online translation portal to help us coordinate all localization effort around SIP Communicator. Just point your favorite browser to:

         http://translate.sip-communicator.org,

create an account and select your project and language to get started.

Today, SIP Communicator supports 10 languages with an average completion rate of 58%. Please help us make these numbers grow and increase the quality of the existing translations. Pootle, the software behind translate.sip-communicator.org, will hopefully allow all of us to rapidly improve the situation.

You always wanted to use SIP Communicator in your native language? With your help, you will soon have no excuse not to do so. Just ask us in case your language is not listed on translate.sip-communicator.org. We need your contribution!

Martin

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#2

Hey Martin,

I've been using Pootle for the last few days and find it extremely
convenient. Thank you for all your efforts! This is definitely going to
help with the internationalization!

Cheers,
Emil

Martin André написа:

···

Hello,

I'm happy to announce that we've setup an online translation portal to
help us coordinate all localization effort around SIP Communicator. Just
point your favorite browser to:

         http://translate.sip-communicator.org,

create an account and select your project and language to get started.

Today, SIP Communicator supports 10 languages with an average completion
rate of 58%. Please help us make these numbers grow and increase the
quality of the existing translations. Pootle, the software behind
translate.sip-communicator.org, will hopefully allow all of us to
rapidly improve the situation.

You always wanted to use SIP Communicator in your native language? With
your help, you will soon have no excuse not to do so. Just ask us in
case your language is not listed on translate.sip-communicator.org. We
need your contribution!

Martin

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#3

It seems that Sip Communicator doesnt support SIP info dialtones.Or, if
it does, it doesnt work right.

In my test setup my Asterisk account's dtmfmode was set "info" and then
"rfc2833" and didnt work in either setup. Other programs (Zoiper,
Sflphone, Linphone) did work. But they have a built-in switch that
permits using either rfc2833 dialtones or SIP info.

Are there plans on implementing a rfc2833/SIP info switch like that?


#4

Hi,

I have just tested in asterisk and with my peer option set to dtmfmode=info
works just fine.

damencho

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On Wed, Mar 10, 2010 at 11:23 AM, Kertesz Laszlo < laszlo.kertesz@infobenefic.ro> wrote:

It seems that Sip Communicator doesnt support SIP info dialtones.Or, if
it does, it doesnt work right.

In my test setup my Asterisk account's dtmfmode was set "info" and then
"rfc2833" and didnt work in either setup. Other programs (Zoiper,
Sflphone, Linphone) did work. But they have a built-in switch that
permits using either rfc2833 dialtones or SIP info.

Are there plans on implementing a rfc2833/SIP info switch like that?


#5

Hmmm. I just tried it again and worked... A few versions back it didnt
work - and it was about one specific IVR, not all.
Thanks and keep up the good work!

regards,

Kertesz Laszlo

···

On Wed, 2010-03-10 at 12:03 +0200, Damian Minkov wrote:

Hi,

I have just tested in asterisk and with my peer option set to
dtmfmode=info works just fine.

damencho

On Wed, Mar 10, 2010 at 11:23 AM, Kertesz Laszlo > <laszlo.kertesz@infobenefic.ro> wrote:
        It seems that Sip Communicator doesnt support SIP info
        dialtones.Or, if
        it does, it doesnt work right.
        
        In my test setup my Asterisk account's dtmfmode was set "info"
        and then
        "rfc2833" and didnt work in either setup. Other programs
        (Zoiper,
        Sflphone, Linphone) did work. But they have a built-in switch
        that
        permits using either rfc2833 dialtones or SIP info.
        
        Are there plans on implementing a rfc2833/SIP info switch like
        that?