date:20041023

Re: [Asterisk-Users] Asterisk, ATA-186 & Sipgate.de / sipgate.co.uk

2004-10-23 Thread BetaTeilchen

Ronald Wiplinger schrieb:
[sipgate.de]
type=friend
username=5552220
secret=my_password3
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.net

should be "fromdomain=sipgate.de"
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=my_password4
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.net

should be "fromdomain=sipgate.co.uk"
nat=yes
context=incomingsipgate
canreinvite=no

extensions.conf:
===
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp; Console interface for demo
[incomingsipgate]
exten => h,1,Hangup
exten => 800,1,Dial(SIP/internestelefon,20,tr)
should be
[incomingsipgate]
exten => 5552220,1,Dial(SIP/internestelefon,20,r)
exten => 4782156,1,Dial(SIP/internestelefon,20,r)

[sipgate.de]
exten => _0049.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten => _0049.,2,Playback(invalid)
exten => _0049.,3,Hangup
should be
(you forgot to number prio 1 !)
exten => _0049.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30,r)
; do not dial international prefix 0049 with Sipgate, if you call from 
same national net !
exten => _0049.,2,Playback(invalid)
exten => _0049.,3,Hangup


[sipgate.co.uk]
exten => _0044.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten => _0044.,2,Playback(invalid)
exten => _0044.,3,Hangup
exten => _0044.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30,tr)
; do not dial international prefix 0044 with Sipgate, if you call from 
same national net !
exten => _0044.,2,Playback(invalid)
exten => _0044.,3,Hangup


I did not understand the paragraph of [incomingsipgate].

This is the context to which all incoming calls from Sipgate will be 
sent to be handled.

I also do not understand EXTEN:1   (should the second phone be EXTEN:2 
???)

please regard correct expression ${EXTEN:1} !
This means "take the variable ($) called {EXTEN} (this is the dialed 
number) and cut the FIRST digit (:1)"
So - if you dial 0049341234567 and then DIAL(SIP/${EXTEN:1}...) this 
will result in dialing 0493411234567 which is not a valid number.

Regards
--
Please visit http://www.ip-phone-forum.de
--
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Re: [Asterisk-Users] KSS/BLF on Asterisk

2004-10-23 Thread Nick Bachmann

[EMAIL PROTECTED] wrote:
Folks,
I am trying to determine the best way to allow a station to monitor the
status of another station.
For example: 
a reception set needing to see the status of 20 or 30 phones
OR
an executive assistant wanting to have appearances of several other
extensions, in order to monitor their status and assist with call
handling.

I know Snom has a phone that you can attach an add-on module to, but I
don't know how you'd program Asterisk to deliver status information to
those buttons.
 

Get a hint! :-)
Check out the "hint" priority in extensions.conf.  There are also some 
details in the wiki.

Nick
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[Asterisk-Users] Asterisk, ATA-186 & Sipgate.de / sipgate.co.uk

2004-10-23 Thread Ronald Wiplinger

I try to get the following to work:
Sipgate.de and sipgate.co.uk are configured as gateway, while the 
ATA-186 has two phone sets attached.

I tried:
ATA settings as described at: 
http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt
(just with a fixed IP)

sip.conf:
==
[general]
context=default 
port=5060   
bindaddr=0.0.0.0
srvlookup=yes   

register => 5552220:[EMAIL PROTECTED]/5552220
register => 4782156:[EMAIL PROTECTED]/4782156
externip = 61.220.121.xx
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
[601]
type=friend
username=601
secret=my_password1
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=601
nat=yes
[602]
type=friend
username=602
secret=my_password2
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=601
nat=yes
[sipgate.de]
type=friend
username=5552220
secret=my_password3
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.net
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=my_password4
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.net
nat=yes
context=incomingsipgate
canreinvite=no

extensions.conf:
===
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
[incomingsipgate]
exten => h,1,Hangup
exten => 800,1,Dial(SIP/internestelefon,20,tr)
[sipgate.de]
exten => _0049.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten => _0049.,2,Playback(invalid)
exten => _0049.,3,Hangup
[sipgate.co.uk]
exten => _0044.,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten => _0044.,2,Playback(invalid)
exten => _0044.,3,Hangup

I did not understand the paragraph of [incomingsipgate].
I also do not understand EXTEN:1   (should the second phone be EXTEN:2 ???)
starting the server with asterisk -vvvcg   brings a lots of lines ;-)
sip show users:
sipgate.co.uk   my_password2  incomingsipgateNo   Alway
sipgate.demy_password1  incomingsipgateNo   Alway
602 my_password4  incomingsipgateNo   Alway
601 my_password3  incomingsipgateNo   Alway
sip show registry:
sipgate.co.uk:5060  4782156   105Registered
sipgate.de:5060   5552220   105Registered

Tests:
601 calls 602busy
00491  busy(1 at sipgate.de should play a tape)
No info on the screen (asterisk: *CLI>  )
What have I forgotten / made wrong?
bye
Ronald
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
version:2.1
end:vcard

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[Asterisk-Users] G.729 on YDL and MacOSX

2004-10-23 Thread Benjamin on Asterisk Mailing Lists

Re: G.729 codec on Yellow Dog Linux for various PPC

Kristian Kielhofner <[EMAIL PROTECTED]> wrote:

> This is probably a good time to ask if there is any
> planned support for a g729 binary for YDL and
> G3/G4, etc. I would love to start playing with
> apple hardware, YDL, and asterisk.
> But I need that binary!

Indeed it is a good time to ask (but always start a new thread ;-)

I have mentioned this before, and I would like to ask EVERYBODY who is
interested to VOICE your interest directly with the respective
vendors. This is the first step and it is VERY IMPORTANT.

I am confindent that an Altivec optimised G.729 codec for the PPC970
CPUs (aka G5) on YDL4 would so clearly trash any Intel or AMD based
system that most serious deployments that require G.729 will end up
using Xserve instead of Intel toyz. Combine this with the fact that
the x86 architecture has hit the wall while IBM is only getting
started. Even Microsoft have recognised the leadership of IBM by going
PPC with their new game console. Before this background it is quite
apparent that there is an interesting market potential for G.729
binaries for LinuxPPC.

However, without requests from customers for a G.729 codec for
LinuxPPC it will take so much longer for an x86 centric shop like
Digium to recognise this potential and consider spending time and
effort on it. Therefore, please, send an email to Digium and tell them
that you want this binary for PPC and continue to nag them about it
again and again and again and again. If as a result, Digium realise
that there is demand, then they will quite possibly provide that
binary.

At the same time, let's also remind TerraSoft
(http://www.terrasoftsolutions.com) that Asterisk on their YDL
platform is alive and that their sponsorship to bring Asterisk to
LinuxPPC was not in vain, that there is finally an opportunity to get
a return on their investment. Let's assume that Digium is simply too
busy with other things and that even if they wanted to, they couldn't
do the G.729 codec for PPC. So, in lieu of Digium providing the codec
for PPC, TerraSoft may recognise the opportunity and step in. But
again, in order for this to happen, it will take requests from
customers.

Therefore, please, send an email to Kai Staats at TerraSoft and tell
them that you'd be very interested to buy G.729 codec binaries for
Asterisk on YDL if they were to offer them, then follow up on that
with reminders to show that you are serious about it. TerraSoft have
been working together with Digium to bring Asterisk to YDL, so there
shouldn't be a problem for the two companies to get together again and
bring the G.729 codec to YDL as well. All it takes for that to happen
is visible customer demand.

Perhaps we should set up some kind of petition page on the Wiki.

Re: G.729 codec on MacOSX for Apple Macintosh

Darren Sessions <[EMAIL PROTECTED]> wrote:

> Or for that matter, is there a planned G729 binary
> for Mac OSX ?

It will probably take a LinuxPPC port first, but here again, why don't
you send email to Apple and tell them that you would rather purchase
oodles of Xserve instead of x86 based servers if only there was a
G.729 codec for OSX. It will take a lot more noise to get Apple to
recognise that there is a market potential than it will take to get
Digium or TerraSoft to do so, but that's no reason not to make a
request.

So, please, send email to Apple and tell them that you have tested
Asterisk on MacOSX -- they have listed our installer on their website
http://www.apple.com/downloads/macosx/unix_open_source/asteriskinstallpackageformacosx.html,
that you found it runs circles around any other product, such as Cisco
Call Manager -- Apple just loves to hear that sort of thing -- and
that the only thing that's missing is the G.729 codec which the open
source community is unable to provide on its own due to the patent
royalties that need to be paid on a reseller-to-patent-holder basis
because there is no end-user-to-patent-holder scheme, that you would
love to buy many many Xserves if Apple was to sell you the missing
codec.

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

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