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

2004-10-24 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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[Asterisk-Users] Asterisk, ATA-186 Sipgate.de / sipgate.co.uk

2004-10-24 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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Re: [Asterisk-Users] KSS/BLF on Asterisk

2004-10-24 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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Re: [Asterisk-Users] Asterisk, ATA-186 Sipgate.de / sipgate.co.uk

2004-10-24 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] Asterisk, ATA-186 Sipgate.de / sipgate.co.uk

2004-10-24 Thread Ronald Wiplinger
BetaTeilchen wrote:
Ronald Wiplinger schrieb:

Thanks for helping me, but it still does not work.

[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)
What is the difference between tr and r ? What does the 20 mean?


[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.

BTW, when I stop Asterisk with stop now I get a
 Yuck! Error in buffer handling ...:
What does this mean?
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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Re: [Asterisk-Users] Asterisk, ATA-186 Sipgate.de / sipgate.co.uk

2004-10-24 Thread BetaTeilchen
Maybe you should start reading here: 
http://www.voip-info.org/wiki-Asterisk+introduction to get basic 
knowledges of Asterisk

Ronald Wiplinger schrieb:
BetaTeilchen wrote:
Ronald Wiplinger schrieb:

Thanks for helping me, but it still does not work.

[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)
What is the difference between tr and r ? What does the 20 mean?


[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.

BTW, when I stop Asterisk with stop now I get a
 Yuck! Error in buffer handling ...:
What does this mean?
bye
Ronald
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[Asterisk-Users] bristtuff segfault

2004-10-24 Thread Jean-Denis Girard
Hi list,
I'd like to have comments from the bristuff / QuadBRI users, others are 
welcome to as I'm really lost and need to move on.

I have the following setup: a first asterisk is connected to the legacy
Alcatel PaBX to connect to a remote site with a second asterisk server.
   PSTN
|
Legacy phones == Alcatel Omnipcx == Asterisk1
|
| IAX
|
  Asterisk2 == 25 SIP phones
Both servers are dell 400sc (Pentium(R) 4 CPU 2.80GHz, 2 IDE disks in
software RAID1). The system has been working flawlessly for a week with 
4 analog lines between the omnipcx and asterisk1 (using TDM04B).

Now I've moved to a 4 BRI link between omnipcx and asterisk1, using 
QuadBRI from Junghanns.net. It does work... for about 3 minutes, and 
then asterisk segfault after hanging up a successful call to the omnipcx.

Backtrace shows the error is in libpri function q931_destroy (q931.c: 
1908), which is coherent with crash after hangup.

Asterisk / Zaptel / Libpri / Qozap on first server have been built with
the scripts from bri-stuff-0.1.0-RC4a.tar.gz. On the card, I only 
changed all groups of 5 jumpers to NT mode. Linux Distro is 
Mandrake-10.0. I tried with kernels (from kernel.org) 2.6.8.1 and 2.4.27 
with and without SMP, and got exactly same results.
(more details below)

Any hints would be much appreciated.
Thanks for a prompt reply.
Jean-Denis
Asterisk1 logs shows warnings on all hangups, eg.
Oct 22 16:39:47 WARNING[1101196208]: PRI: Can't destroy call 133!
Oct 22 16:39:47 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1
Oct 22 16:39:54 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1
Oct 22 16:39:58 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1
Oct 22 16:40:04 WARNING[1101196208]: PRI: Can't destroy call 135!
Oct 22 16:40:04 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1
Oct 22 16:40:41 WARNING[1101196208]: PRI: Can't destroy call 137!
Oct 22 16:40:41 WARNING[1101196208]: Hangup on bad channel 0/2 on span 1
Oct 22 16:41:09 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1
Oct 22 16:41:13 WARNING[1101196208]: Hangup on bad channel 0/1 on span 1
Oct 22 16:41:19 NOTICE[1121725360]: I should never be called!
Oct 22 16:41:53 WARNING[1101196208]: PRI: Can't destroy call 140!
Oct 22 16:41:53 WARNING[1101196208]: Hangup on bad channel 0/2 on span 1
Loading qozap shows no error except for devfs, but I doubt this is the
source of the problem:
Oct 22 16:29:43 asterisk1 kernel: Zapata Telephony Interface Registered
on major 196
Oct 22 16:29:56 asterisk1 kernel: PCI: Enabling device 02:02.0 ( -
0003)
Oct 22 16:29:56 asterisk1 kernel: PCI: Found IRQ 10 for device 02:02.0
Oct 22 16:29:56 asterisk1 kernel: PCI: Sharing IRQ 10 with 00:1f.3
Oct 22 16:29:56 asterisk1 kernel: PCI: Sharing IRQ 10 with 00:1f.5
Oct 22 16:29:56 asterisk1 kernel: qozap: Junghanns.NET quadBRI card
configured at mem 0xe08e IRQ 10 HZ 100 CardID 0
Oct 22 16:29:56 asterisk1 kernel: qozap: S/T ports: 4 [ NT NT NT NT ]
Oct 22 16:29:56 asterisk1 kernel: card 1 span 1 state G0 (A_ST_RD_STA = 0x0)
Oct 22 16:29:56 asterisk1 kernel: card 1 span 2 state G0 (A_ST_RD_STA = 0x0)
Oct 22 16:29:56 asterisk1 kernel: card 1 span 3 state G0 (A_ST_RD_STA = 0x0)
Oct 22 16:29:56 asterisk1 kernel: card 1 span 4 state G0 (A_ST_RD_STA = 0x0)
Oct 22 16:29:56 asterisk1 kernel: qozap: 1 multiBRI card(s) in this box,
4 BRI ports total.
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/1 to /dev/zap/1
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/2 to /dev/zap/2
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/3 to /dev/zap/3
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/4 to /dev/zap/4
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/5 to /dev/zap/5
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/6 to /dev/zap/6
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/7 to /dev/zap/7
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/8 to /dev/zap/8
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/9 to /dev/zap/9
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/10 to /dev/zap/10
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/11 to /dev/zap/11
Oct 22 16:29:56 asterisk1 devfsd[168]: error copying:
/lib/dev-state/zap/12 to /dev/zap/12
Oct 22 16:30:36 asterisk1 kernel: Registered tone zone 2 (France)
Oct 22 16:30:36 asterisk1 kernel: card 1 span 1 state G2 (A_ST_RD_STA =
0x82)
Oct 22 16:30:36 asterisk1 kernel: card 1 span 2 state G2 (A_ST_RD_STA =
0xc2)
Oct 22 16:30:36 asterisk1 kernel: card 1 span 3 state G2 (A_ST_RD_STA =
0x82)
Oct 22 16:30:36 asterisk1 kernel: card 1 span 4 state G2 (A_ST_RD_STA =
0xc2)
Oct 22 

Re: [Asterisk-Users] iLBC/PCM16 Huge Cost

2004-10-24 Thread Trevor Peirce
Todd Lieberman wrote:
Wo trevor, Format and start over?  Don't go crazy, just remove the files
created by make install.
Fighting for weeks to get a more-or-less stable telephone system can 
drive a man to do extraordinary things like rebuilding a server from 
scratch!

We are making process, however.  With the 0.59r mpg123, I see no 
processes consuming all sorts of CPU power once * has been running for a 
few hours.

show translation still reveals the iLBC column in the 8700 to 9600 range 
though.  LPC10's row is also in the 900s.

show translation recalc 10 still causes the * console to stop responding 
as well.

When I first bootup *, it consumes nearly 100% of CPU, all in user... 
system is less than 1%.  The time that it monopolizes the processor 
varies as well.
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Re: [Asterisk-Users] Hardware

2004-10-24 Thread Tzafrir Cohen
On Sat, Oct 23, 2004 at 06:40:12PM -0600, Michael Loftis wrote:
 
 
 --On Saturday, October 23, 2004 19:56 -0400 Stan Brinkerhoff 
 [EMAIL PROTECTED] wrote:
 
 Look for support by whatever operating system you plan on running.
 
 I second thatpretty much any P4 based hardware should be perfectly fine 
 for asterisk.  I'd tend to lean towards SCSI drives though, but other than 
 that go to town!

Why scsi?

I thought that Asterisk doesn't have much disk IO. At least that this is
not a bottleneck.

-- 
Tzafrir Cohen   +---+
http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
mailto:[EMAIL PROTECTED]   +---+
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Re: [Asterisk-Users] Hardware

2004-10-24 Thread christophe de coninck




I would use a Western Digital Raptor SATA Harddisk, also gives you a performance boost of your system + it aint that expensive as scsi.

And my dream setup for asterisk would be:
dual xeon, intel xeon motherboard, 2gig ram for each cpu and a few raptor or scsi disks + some wildcard digium telephony cards to call with 10users at a time to a normal phone number.

On Sun, 2004-10-24 at 11:49, Tzafrir Cohen wrote:

On Sat, Oct 23, 2004 at 06:40:12PM -0600, Michael Loftis wrote:
 
 
 --On Saturday, October 23, 2004 19:56 -0400 Stan Brinkerhoff 
 [EMAIL PROTECTED] wrote:
 
 Look for support by whatever operating system you plan on running.
 
 I second thatpretty much any P4 based hardware should be perfectly fine 
 for asterisk.  I'd tend to lean towards SCSI drives though, but other than 
 that go to town!

Why scsi?

I thought that Asterisk doesn't have much disk IO. At least that this is
not a bottleneck.




-- 
Christophe De Coninck | Zarek K 

http://www.zarekk.be
mailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED]








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Re: [Asterisk-Users] Fedora 2, Kudzu and X100P

2004-10-24 Thread Tzafrir Cohen
On Sat, Oct 23, 2004 at 09:29:00PM -0500, Carlos Chavez wrote:
  I am installing a new * server using Fedora Core 2 but I ran into a
 problem after I installed the X100P.  When FC2 boots it runs KUDZU to detect
 new hardware and it detected the card and insists on loading the module
 crc_ccitt before the zaptel module.  Because of this I cannot load the wcfxo
 module without the computer crashing.  I have already erase the entry in
 /etc/sysconfig/hwconf and turned kudzu off during boot.
 
  Anyone know of a way to fix this (short or reinstalling FC2)?

One obvious solution is not to automatically load kudzu.

  chkconfig --remove kudzu

Another obvious solution of the same sort is modprobing the zaptel
module earlier in the boot process. 

I can't seem to figure out , though, where kudzu takes its modue names
from. I haven't bothred reading th source yet, though (not from
/usr/share/hwdata, it seems)

-- 
Tzafrir Cohen   +---+
http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend|
mailto:[EMAIL PROTECTED]   +---+
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[Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Benjamin on Asterisk Mailing Lists
A customer has ordered some voice prompts from Digium's TheVoice
online store. They say the recordings' sound was good when they
listened to it on their Windoze boxes. However, then Asterisk is
playing back the recordings, the volume is far too high and they sound
really bad. This is particularly noticeable since the IVR menu mixes
those ordered recordings with recordings that are already part of the
Asterisk distribution. The volume of the included recordings are much
lower and they sound much better than the ordered recordings.

I wonder why Digium would deliver recordings that differ so much from
the included set of recordings.

However, the format the customer ordered was WAV, whereas all the
included recordings are of course GSM. Has anybody had similar
experiences? I tried to convert the WAV files to GSM using sox but
since I don't know what parameters are best in this case, the results
weren't satisfactory. Any suggestions?

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

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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Steve Totaro
Things that are productive.

1.  I am sure there are free programs that will allow you to adjust the
files to sound more like the originial recordings as well as converting them
to gsm.  Do some searching and learning and fix it yourself.  Mail Digium
directly so that they are aware of the problem and can correct it for future
recordings.  Possibly something like this
http://www.softpicks.net/software/Complete-Audio-Converter-Lite-1470.htm

2 .  If you dont want to go through all of that, kindly ask Digium to have
the files fixed for you.  I seriously doubt they have their own sound stage
and most likely outsource this type of business.  Chances are the people
they outsource the business to are experts and have sophisticated equipment
to do this with very fast turn around time.

3.  Email thousands of people that will probably ignore you or not know the
answer (with the exception of myself).

I would choose number one.

Thanks,
Steve


- Original Message - 
From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 6:37 AM
Subject: [Asterisk-Users] Digium TheVoice recordings' sound terrible


 A customer has ordered some voice prompts from Digium's TheVoice
 online store. They say the recordings' sound was good when they
 listened to it on their Windoze boxes. However, then Asterisk is
 playing back the recordings, the volume is far too high and they sound
 really bad. This is particularly noticeable since the IVR menu mixes
 those ordered recordings with recordings that are already part of the
 Asterisk distribution. The volume of the included recordings are much
 lower and they sound much better than the ordered recordings.

 I wonder why Digium would deliver recordings that differ so much from
 the included set of recordings.

 However, the format the customer ordered was WAV, whereas all the
 included recordings are of course GSM. Has anybody had similar
 experiences? I tried to convert the WAV files to GSM using sox but
 since I don't know what parameters are best in this case, the results
 weren't satisfactory. Any suggestions?

 thanks
 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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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Steve Totaro
Option #4, send me the files and $100 via paypal and I will fix them for
you.


- Original Message - 
From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 6:37 AM
Subject: [Asterisk-Users] Digium TheVoice recordings' sound terrible


 A customer has ordered some voice prompts from Digium's TheVoice
 online store. They say the recordings' sound was good when they
 listened to it on their Windoze boxes. However, then Asterisk is
 playing back the recordings, the volume is far too high and they sound
 really bad. This is particularly noticeable since the IVR menu mixes
 those ordered recordings with recordings that are already part of the
 Asterisk distribution. The volume of the included recordings are much
 lower and they sound much better than the ordered recordings.

 I wonder why Digium would deliver recordings that differ so much from
 the included set of recordings.

 However, the format the customer ordered was WAV, whereas all the
 included recordings are of course GSM. Has anybody had similar
 experiences? I tried to convert the WAV files to GSM using sox but
 since I don't know what parameters are best in this case, the results
 weren't satisfactory. Any suggestions?

 thanks
 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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[Asterisk-Users] Call Waiting

2004-10-24 Thread Nikhil Jogia
Hi,

I have just set up an Asterisk box.it sure is a big job to get
everything perfect, especially when you have picky users.

Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
connected to the LAN.

1 of the lines connected to the X100P's goes straight to extension 1000
after a short greeting.

Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext
1000 hears a small beep every few seconds. This is obviously call
waiting.

My question is how do I answer that incoming call whilst on a call? I
have looked around, tried *0 and even 0*, the flash key, but to no avail
:(



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Re: [Asterisk-Users] Digium Wildcard T1 Compatibility

2004-10-24 Thread Steve Totaro
A flex grow is like a channel bank.  A normal PRI comes into a router.  The
router breaks out some channels for data and the other voice channels become
analog POTS lines.  You will need POTS cards.

I am positive that you could have your T100P and asterisk provide this
function so that you wouldnt need their equipment or POTS.  Just depends on
the tech you get whether they will help or not.  Just read the first
paragraph of the product description on Digium's site.
http://www.digium.com/index.php?menu=wildcard_t100p


- Original Message - 
From: Cirelle Enterprises [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, October 22, 2004 4:06 PM
Subject: Re: [Asterisk-Users] Digium Wildcard T1 Compatibility



- Original Message - 
From: Daniel Daley [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 21, 2004 5:49 PM
Subject: [Asterisk-Users] Digium Wildcard T1 Compatibility


| Hi,
|
| I have a quick question about the T100P. I've used the card before in a
| PRI setup and it worked great. I'm now trying to figure out a setup for
| another company that gets services from Verizon. They offer what they
| call a flexgrow T1 where they say the voice lines are delivered as just
| standard POTS channels. Will the wildcard handle this kind of T1 or is
| that something you would need to break out into separate lines and go
| into POTS cards?
|
| Thanks,
|
| --Dan--
|



for what it's worth, we were told to use RJ48C (Std Ethernet Cable)
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[Asterisk-Users] chan_sip CallerPres support?

2004-10-24 Thread Roy Sigurd Karlsbakk
hi
would it be hard to implement CallerPres support in chan_sip?
roy
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Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)

2004-10-24 Thread Steve Totaro
May be risky if your email is screwy but it solves your problem

Add:  delete=yes in your voicemail.conf.


- Original Message - 
From: Stephen R. Besch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, October 22, 2004 11:55 AM
Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue)


 Todd Routhier - Lightwave Technologies, LLC. wrote:
  OK, this is a different flashing issue than the one that's being talked
  about.
 
  I have a few of these phones (GrandStream 101) and when a voicemail is
  received the light on the LED starts blinking and the dial tine
stutters,
  this is cool. BUT. How the  do I get it to stop, I have had mine
  covered with paper for the last 2 days because the blinking LED panel is
  driving me nuts. I have received the messages in my email and looked at
them
  on the web.
 
  I am thinking that I have to check them by calling into the Asterisk
system
  and mark them as read or something in order for this to quite.
 
  Two problems, I don't know how to check them by phone just yet and I
will
  likely never check them by phone.
 
 You absolutely must get the message file deleted from the mail store. It
 doesn't matter how you delete the message either. You can use the *
 phone interface or simply delete the files associated with the message.
 As soon as the files are gone from the INBOX, * sends a SIP command to
 the phone to turn off the MWI. As long as there are any active messages
 in the INBOX, the light stays on.

 If you would like to be a guinea pig for my VB program that allows you
 to manage your mail folders and messages from a Windows GUI, I'll send
 you a copy. The only caveat is that I haven't yet found a way to get
 perfect synchronization with file access to the mail store. The user
 needs to be careful not to modify the INBOX while * is taking a new
 message. For me this is usually not a problem, since I would never be
 not answering the phone when I am listening to/moving/deleting messages
 - as a result, * would never be writing a new message when I was using
 the VB program.

 The program requires the VB6 runtime (available from MS for free) and
 that you have SAMBA running on the * server.

 Sincerely,

 Stephen R. Besch

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[Asterisk-Users] Problem compiling ZPAHFC with Suse 9.1, Kernel 2.6.5

2004-10-24 Thread Joachim Grübler
 
Hi,
 
 I've some problems compiling/installing the ZAPHFC-Driver. I've download the actuell 
version bristuff-0.1.0-RC4a from junghanns.net. I use SUSE 9.1 with kernel 2.6.5.-111. 
I've made the symbolic link to Linux-2.6 and test the link successfully. I've done 
make oldconig, make menuconfig and make in the linux-directory.

When I start ./compile.sh in the bri-stuff-directory (./download.sh alraedy done 
before), zaptel and libpri will be compiled without problems. But compiling of qozap 
and zaphfc will end wit error: zt_register, -_unregister, -_transmit, -_receive and 
-_chunk are not defined

It's possible to install the ZAPHFS-driver with make loadNT but it reports 0 
channels configured.

I've alraedy googled this problem but find only users with the same problem, no 
resolution. Have anyone one?



Joachim



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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Benjamin on Asterisk Mailing Lists
Hi Steve,

On Sun, 24 Oct 2004 07:47:17 -0400, Steve Totaro
[EMAIL PROTECTED] wrote:

 1.  I am sure there are free programs that will allow you to adjust the
 files to sound more like the originial recordings as well as converting them
 to gsm.

that's all very cool, but if you read my post carefully -- you did
read more than just the subject line, did you not?! -- then you will
find that I had mentioned I was unable to get acceptable results using
sox because I don't know what parameters to use.

Are you suggesting that sox is not the right tool for the job, no
matter what parameters? I would be very surprised if that was the
case.

Since I didn't think this is the only time that Digium delivered WAV
recordings that are out of sync with the Asterisk GSM sound library,
in other words, that this was a known problem, I honestly expected
that somebody else had already discovered what parameters to use with
sox to do this conversion properly.

 Do some searching and learning and fix it yourself.

I am sure you will appreciate that not everybody is a sound engineer
and not everybody has the time to spend more than a full day of
experimenting to fix five recordings. You will also appreciate that in
such cases people go to mailing lists like this one where they can
reasonably expect that somebody else has already discovered what the
proper parameters are.

Last but not least, my dear friend, if you had the courtesy to browse
the list archive and check out how many times I came here for help
versus how many times I have helped others, then you will probably
find that your remarks are misplaced.

 Digium directly so that they are aware of the problem and can correct it for future
 recordings.

Done that already. No response. Hence my posting here.

 3.  Email thousands of people that will probably ignore you or not know the
 answer (with the exception of myself).

Well, if you know the answer, then why do you have to write a novel
instead of posting what -- from your experience -- the correct sox
parameters should look like, for example?

 I would choose number one.

And quite possibly revert to asking the list after you have failed to
produce any acceptable results and Digium didn't respond to you.

But then again, I feel sorry that you had such a bad day today. While
I am glad that your wrath didn't hit some poor newbie who would
probably be scared away from ever posting to this list again, I
sincerely hope that you feel better now.

kind regards
benjamin

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

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[Asterisk-Users] How to create Groups/members and do Conferencing?

2004-10-24 Thread Smarty
 Hi, I have just been able to compile asterisk, so that says that I'm fairnly new to Asterisk. I'm still figuring out how to use it with my User Agents. My requirement is:1. To make a "Group" containing some agents (SIP User Agents) as members.2. To start a conference between the members of the Group. (The asterisk server shoulddo the conferencing between these SIP User Agents. So the asterisk server shouldbe able to understand the request from one member to be redirected to all the group members).I'll be really grateful if I could have some suggestions as to how can I create the Group and start the conference?Note: The SIP User agents and the asterisk server are running on the same machine. Join Excite! - http://www.excite.comThe most personalized portal on the Web!
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Re: [Asterisk-Users] Call Waiting

2004-10-24 Thread Steve Totaro
You are supposed to be able to either press flash or quickly push the actual
hook switch.


- Original Message - 
From: Nikhil Jogia [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 7:54 AM
Subject: [Asterisk-Users] Call Waiting


 Hi,

 I have just set up an Asterisk box.it sure is a big job to get
 everything perfect, especially when you have picky users.

 Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
 connected to the LAN.

 1 of the lines connected to the X100P's goes straight to extension 1000
 after a short greeting.

 Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext
 1000 hears a small beep every few seconds. This is obviously call
 waiting.

 My question is how do I answer that incoming call whilst on a call? I
 have looked around, tried *0 and even 0*, the flash key, but to no avail
 :(



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RE: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Kevin Walsh
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
 A customer has ordered some voice prompts from Digium's TheVoice
 online store. They say the recordings' sound was good when they
 listened to it on their Windoze boxes. However, then Asterisk is
 playing back the recordings, the volume is far too high and they sound
 really bad. This is particularly noticeable since the IVR menu mixes
 those ordered recordings with recordings that are already part of the
 Asterisk distribution. The volume of the included recordings are much
 lower and they sound much better than the ordered recordings.
 
 I wonder why Digium would deliver recordings that differ so much from
 the included set of recordings.

Did you ask Digium?  There's probably not a lot of point in asking
for customer support from any specific vendor in this mail list.
You should contact the vendor's support department directly.

 
 However, the format the customer ordered was WAV, whereas all the
 included recordings are of course GSM. Has anybody had similar
 experiences? I tried to convert the WAV files to GSM using sox but
 since I don't know what parameters are best in this case, the results
 weren't satisfactory. Any suggestions?
 
You could try using sox.  That will convert WAV files to GSM and adjust
the volume for you during the conversion process.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Brian Roy
 However, the format the customer ordered was WAV, whereas all the
 included recordings are of course GSM. Has anybody had similar
 experiences? I tried to convert the WAV files to GSM using sox but
 since I don't know what parameters are best in this case, the results
 weren't satisfactory. Any suggestions?

Benjamin,

Don't know if this helps you or not, but this is taken right from
Jtodd's wiki page.

**my disclaimer** Might want to backup your sounds before doing this though.

#!/bin/sh 
 tmpfile=/tmp/rescale$$.wav 
 for i in *.wav; do  
   scale=$(sox $i /tmp/foo.wav stat -v 21) 
   if [ $scale != 1.000 ]; then 
 echo -n Rescale $i... 
 cp $i $tmpfile 
 sox $tmpfile -v $scale $i 
 echo 
   fi 
 done 

The wiki page is at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files

Hope this helps,

-Chuji
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[Asterisk-Users] Re: Direct SIP connection to Vonage service

2004-10-24 Thread Stewart Nelson
Hi Benjamin,
I looked at NuFone.net and some others, but it appears that
IAX is not right for my system.

I'd say this is only because you don't know enough about IAX yet ;-)

[Many comments explaining how IAX would work wonderfully if all my
 VoIP hardware were replaced with IAX-compatible equipment]
No, I don't want to replace existing gear.  It would be expensive,
disrupt operations, take lots of time to set up, and I don't want
the administrative hassle of running multiple Asterisk systems
for such a small network.
There are other reasons, too.  For example, the Cisco 827-4V is very
reliable, because it has no hard drive and no fans.  If *your*
Asterisk system fails, you can zip over to Akihabara and get what
you need, even on Sunday.  Rue Montgallet is not the same!
IAX is probably the ideal protocol for an interoffice trunk carrying
many calls at once, but for me, it seems better to gradually migrate
to a SIP-based system, with a single Asterisk server in Reno, and
retaining present hardware.
If NuFone service is reliable, good quality, competitively priced,
and they also support the open source community, they'll get my
business.  But I'll connect via SIP, so existing equipment can be
better utilized.
When I'm in a hotel, stuck behind a NAT over which I have no control,
sure, I'll use IAX to connect to the server (and tolerate the media
proxy delays in that case.)
Have I missed something?
Regards,
Stewart
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RE: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Kevin Walsh
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
 On Sun, 24 Oct 2004 07:47:17 -0400, Steve Totaro [EMAIL PROTECTED] wrote:
  1.  I am sure there are free programs that will allow you to adjust the
  files to sound more like the originial recordings as well as converting
  them 
  to gsm.
 
 that's all very cool, but if you read my post carefully -- you did
 read more than just the subject line, did you not?! -- then you will
 find that I had mentioned I was unable to get acceptable results using
 sox because I don't know what parameters to use.

It looks as if I didn't read your article carefully enough either.
You can ignore the followup I just posted.

-- 
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  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
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_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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[Asterisk-Users] random crash at startup

2004-10-24 Thread Tomas Carnecky
Hi folks,
I have upgraded asterisk from 0.8 to 1.0 on my gentoo server and it 
won't start now. It crashes on random points while loading the modules 
somewhere between res_crypto and chan_iax2

the last messages are either:
===
Yuck! Error in buffer handling...: Success
Asterisk cleanly ending (2).
===
 [app_random.so] = (Random goto)
  == Registered application 'Random'
 [app_transfer.so]Yuck! Error in buffer handling...: Broken pipe
===
 [chan_zap.so] = (Zapata Telephony w/PRI)
Asterisk cleanly ending (2).
===
 [res_crypto.so] = (Cryptographic Digital Signatures)
Found new ID3 Header
Beginning asterisk shutdown
Executing last minute cleanups
  == Destroying any remaining musiconhold processes
Asterisk cleanly ending (2).
===
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [res_musiconhold.so] = (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
Oct 24 15:44:28 WARNING[1077071984]: res_musiconhold.c:561 moh_register: 
Unable to open pseudo channel for timing...  Sound may be choppy.
Found new ID3 Header
Warning, flexible rate not heavily tested!
Beginning asterisk shutdown
Executing last minute cleanups
Asterisk cleanly ending (2).
Ouch ... error while writing audio data: : Broken pipe
===
 [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
Oct 24 15:44:49 WARNING[1077071984]: chan_iax2.c:7409 load_module: 
Unable to open IAX timing interface: No such device or address
  == Manager registered action IAXpeers
  == Parsing '/etc/asterisk/iax.conf': Found
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver 
(Ver 2))
  == Using TOS bits 16
  == IAX Ready and Listening on 0.0.0.0 port 4569
Yuck! Error in buffer handling...: Success
Asterisk cleanly ending (2).
===
 [chan_sip.so] = (Session Initiation Protocol (SIP))
  == Parsing '/etc/asterisk/sip.conf': Found
-- SIP Seeding 'janat' at [EMAIL PROTECTED]:5060 for 1800
Beginning asterisk shutdown
Executing last minute cleanups
  == Destroying any remaining musiconhold processes
Yuck! Error in buffer handling...: Broken pipe
Asterisk cleanly ending (2).
===
and so on...

what is it?
tom
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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Elliot Moore
Helpful URLS about SOX/wav/gsm
Have you seen these?
Converting:
http://www.voip-info.org/wiki- 
Convert+WAV+audio+files+for+use+in+Asterisk

Volume:
http://www.voip-info.org/wiki-Asterisk+sound+files
Other bits and bobs:
http://www.marko.net/asterisk/archives/0212/0384.html
e.

On 24 Oct 2004, at 11:37, Benjamin on Asterisk Mailing Lists wrote:
A customer has ordered some voice prompts from Digium's TheVoice
online store. They say the recordings' sound was good when they
listened to it on their Windoze boxes. However, then Asterisk is
playing back the recordings, the volume is far too high and they sound
really bad. This is particularly noticeable since the IVR menu mixes
those ordered recordings with recordings that are already part of the
Asterisk distribution. The volume of the included recordings are much
lower and they sound much better than the ordered recordings.
I wonder why Digium would deliver recordings that differ so much from
the included set of recordings.
However, the format the customer ordered was WAV, whereas all the
included recordings are of course GSM. Has anybody had similar
experiences? I tried to convert the WAV files to GSM using sox but
since I don't know what parameters are best in this case, the results
weren't satisfactory. Any suggestions?
thanks
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
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Re: [Asterisk-Users] random crash at startup

2004-10-24 Thread Steve Totaro
Did you upgrade zaptel and libpri before upgrading asterisk?


- Original Message - 
From: Tomas Carnecky [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 9:44 AM
Subject: [Asterisk-Users] random crash at startup


 Hi folks,
 
 I have upgraded asterisk from 0.8 to 1.0 on my gentoo server and it 
 won't start now. It crashes on random points while loading the modules 
 somewhere between res_crypto and chan_iax2
 
 the last messages are either:
 ===
 Yuck! Error in buffer handling...: Success
 Asterisk cleanly ending (2).
 ===
   [app_random.so] = (Random goto)
== Registered application 'Random'
   [app_transfer.so]Yuck! Error in buffer handling...: Broken pipe
 ===
   [chan_zap.so] = (Zapata Telephony w/PRI)
 Asterisk cleanly ending (2).
 ===
   [res_crypto.so] = (Cryptographic Digital Signatures)
 Found new ID3 Header
 Beginning asterisk shutdown
 Executing last minute cleanups
== Destroying any remaining musiconhold processes
 Asterisk cleanly ending (2).
 ===
 Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
   [res_musiconhold.so] = (Music On Hold Resource)
== Parsing '/etc/asterisk/musiconhold.conf': Found
 Oct 24 15:44:28 WARNING[1077071984]: res_musiconhold.c:561 moh_register: 
 Unable to open pseudo channel for timing...  Sound may be choppy.
 Found new ID3 Header
 Warning, flexible rate not heavily tested!
 Beginning asterisk shutdown
 Executing last minute cleanups
 Asterisk cleanly ending (2).
 Ouch ... error while writing audio data: : Broken pipe
 ===
   [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
 Oct 24 15:44:49 WARNING[1077071984]: chan_iax2.c:7409 load_module: 
 Unable to open IAX timing interface: No such device or address
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
== Registered channel type 'IAX2' (Inter Asterisk eXchange Driver 
 (Ver 2))
== Using TOS bits 16
== IAX Ready and Listening on 0.0.0.0 port 4569
 Yuck! Error in buffer handling...: Success
 Asterisk cleanly ending (2).
 ===
   [chan_sip.so] = (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
  -- SIP Seeding 'janat' at [EMAIL PROTECTED]:5060 for 1800
 Beginning asterisk shutdown
 Executing last minute cleanups
== Destroying any remaining musiconhold processes
 Yuck! Error in buffer handling...: Broken pipe
 Asterisk cleanly ending (2).
 ===
 and so on...
 
 what is it?
 
 tom
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[Asterisk-Users] Error when compiling asterisk-oh323

2004-10-24 Thread Willis Wang
When I try to compile asterisk-oh323, errors as following: 

for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make[1]: Entering directory `/root/willis/asterisk-oh323-0.6.3b/wrapper'
./check_ver /root/willis/pwlib pwlib
./check_ver /root/willis/openh323 openh323
gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o 
asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o
make[1]: Leaving directory `/root/willis/asterisk-oh323-0.6.3b/wrapper'
make[1]: Entering directory 
`/root/willis/asterisk-oh323-0.6.3b/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE 
-I/root/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c
chan_oh323.c: In function `oh323_call':
chan_oh323.c:1237: structure has no member named `callerid'
chan_oh323.c:1239: structure has no member named `callerid'
chan_oh323.c:1241: structure has no member named `callerid'
chan_oh323.c: In function `ast_oh323_new':
chan_oh323.c:2381: structure has no member named `dnid'
chan_oh323.c:2391: structure has no member named `callerid'
chan_oh323.c:2392: structure has no member named `callerid'
chan_oh323.c:2393: structure has no member named `callerid'
chan_oh323.c:2398: structure has no member named `callerid'
chan_oh323.c:2399: structure has no member named `callerid'
chan_oh323.c:2400: structure has no member named `callerid'
chan_oh323.c:2402: structure has no member named `callerid'
chan_oh323.c:2407: structure has no member named `callerid'
chan_oh323.c:2408: structure has no member named `callerid'
chan_oh323.c:2410: structure has no member named `callerid'
chan_oh323.c:2412: structure has no member named `callerid'
chan_oh323.c:2416: structure has no member named `callerid'
chan_oh323.c:2419: structure has no member named `ani'
chan_oh323.c:2419: structure has no member named `callerid'
chan_oh323.c:2425: structure has no member named `callerid'
chan_oh323.c:2426: structure has no member named `callerid'
chan_oh323.c: In function `load_module':
chan_oh323.c:4697: warning: passing arg 4 of `ast_channel_register' from 
incompatible pointer type
make[1]: *** [chan_oh323.o] Error 1
make[1]: Leaving directory 
`/root/willis/asterisk-oh323-0.6.3b/asterisk-driver'
make: *** [subdirs_all] Error 1 

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Re: [Asterisk-Users] random crash at startup

2004-10-24 Thread Tomas Carnecky
Steve Totaro wrote:
Did you upgrade zaptel and libpri before upgrading asterisk?
do I need zaptel?
I have libpri-1.0.0 but no zaptel installed.
in the gentoo ebuild the dependecy is like thik:
DEPEND=virtual/libc
media-sound/mpg123
dev-libs/newt
doc? ( app-doc/doxygen )
alsa? ( media-libs/alsa-lib )
mysql? ( dev-db/mysql )
gtk? ( =x11-libs/gtk+-1.2* )
!nopri? ( =net-libs/libpri-1.0.0 )
!nozaptel? ( =net-misc/zaptel-1.0.0
 =net-libs/zapata-1.0.0 )
and I did
USE=~doc ~alsa ~mysql ~gtk nopri nozaptel emerge asterisk
do I need alsa, mysql, libpri or zaptel?
tom
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[Asterisk-Users] (iax|sip)friends in extconfig?

2004-10-24 Thread Roy Sigurd Karlsbakk
hi
I'm currently using sipfriends from asterisk-stable and I've enabled 
MYSQL_USERS as well. Are mysql/odbc/whatever _users_ available in 
extconfig yet?

roy
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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Andrew Kohlsmith
On October 24, 2004 07:47 am, Steve Totaro wrote:
 2 .  If you dont want to go through all of that, kindly ask Digium to have
 the files fixed for you.  I seriously doubt they have their own sound stage
 and most likely outsource this type of business.  Chances are the people
 they outsource the business to are experts and have sophisticated equipment
 to do this with very fast turn around time.

Actually The Voice is Alison, and she does work at Digium.  

-A.
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Re: [Asterisk-Users] iLBC/PCM16 Huge Cost

2004-10-24 Thread Andrew Kohlsmith
On October 23, 2004 10:58 pm, Michael Loftis wrote:
 mmm... any packaging is better than none.  I regularly destroy things on
 systems when it's not been put into proper packaging because we upgrade the
 system, and there's no record of something being installed, nor what it
 depends on, so it gets broken.

http://asic-linux.com.mx/~izto/checkinstall/

Checkinstall is your friend.  RPM, DEB and Slackware .TGZ formats supported.

-A.
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[Asterisk-Users] Re: chan_sip CallerPres support?

2004-10-24 Thread Andreas Anderson

would it be hard to implement CallerPres support in chan_sip?
There is support for outgoing calls, but this patch breakes incoming 
callerid:

http://bugs.digium.com/bug_view_page.php?bug_id=0002471
Greez
Andreas
_
Listen to music online with the Xtra Broadband Channel  
http://xtra.co.nz/broadband

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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Steve Totaro
I know she works at Digium but they probably go down the street to a real
sound stage to do the recordings via 3rd party.

A sound stage is a facility used to create and process professional
recordings.  They can be used by anyone employed by an company.


- Original Message - 
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 10:05 AM
Subject: Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible


 On October 24, 2004 07:47 am, Steve Totaro wrote:
  2 .  If you dont want to go through all of that, kindly ask Digium to
have
  the files fixed for you.  I seriously doubt they have their own sound
stage
  and most likely outsource this type of business.  Chances are the people
  they outsource the business to are experts and have sophisticated
equipment
  to do this with very fast turn around time.

 Actually The Voice is Alison, and she does work at Digium.

 -A.
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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Joe Greco
 I know she works at Digium but they probably go down the street to a real
 sound stage to do the recordings via 3rd party.
 
 A sound stage is a facility used to create and process professional
 recordings.  They can be used by anyone employed by an company.

http://www.theivrvoice.com/

would seem to imply otherwise.  I'd be a bit surprised if any company had
enough work to keep her employed full-time, so the works at Digium line
sounds a bit fishy to me.

Regards,

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Andrew Kohlsmith
On October 24, 2004 10:24 am, Steve Totaro wrote:
 I know she works at Digium but they probably go down the street to a real
 sound stage to do the recordings via 3rd party.

Oh I dunno, for telephone IVR you don't need much of a sound stage.  Convert a 
bathroom into one with a lot of insulation, thick carpet and soft walls.  
Throw in a good mic, a decent preamp and mixer to adjust levels and that's 
about all you'd need for doing good quality IVR voicing.

It's not like this is going on to a CD or something.  :-)

-A.
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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Benjamin on Asterisk Mailing Lists
On Sun, 24 Oct 2004 14:47:56 +0100, Elliot Moore
[EMAIL PROTECTED] wrote:
 Helpful URLS about SOX/wav/gsm
 Have you seen these?

[snip URLs]

yes, I have played with those and all I did achieve was making the
recordings worse, but thanks anyway.

However, it seems now that this is not a common problem so I have to
check back with the customer whether these files are really the
originals. You never know, they might have messed around with those.
If anything else fails I will ask them to reorder the prompts once
more and get them delivered as GSM.

thanks again
rgds
benjk

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

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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Benjamin on Asterisk Mailing Lists
On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote:
 http://www.theivrvoice.com/
 
 would seem to imply otherwise.  I'd be a bit surprised if any company had
 enough work to keep her employed full-time, so the works at Digium line
 sounds a bit fishy to me.

I think when he wrote 'She does work at Digium' it was meant in the
sense of She is doing work at Digium', or 'She does (some) work for
Digium ;-)

rgds
benjk

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

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Re: [Asterisk-Users] random crash at startup

2004-10-24 Thread Steve Totaro
get the new versions of libpri zaptel and asterisk and install them in that
order.  should work.


- Original Message - 
From: Tomas Carnecky [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 10:05 AM
Subject: Re: [Asterisk-Users] random crash at startup


 Steve Totaro wrote:
  Did you upgrade zaptel and libpri before upgrading asterisk?
 

 do I need zaptel?

 I have libpri-1.0.0 but no zaptel installed.

 in the gentoo ebuild the dependecy is like thik:
 DEPEND=virtual/libc
  media-sound/mpg123
  dev-libs/newt
  doc? ( app-doc/doxygen )
  alsa? ( media-libs/alsa-lib )
  mysql? ( dev-db/mysql )
  gtk? ( =x11-libs/gtk+-1.2* )
  !nopri? ( =net-libs/libpri-1.0.0 )
  !nozaptel? ( =net-misc/zaptel-1.0.0
   =net-libs/zapata-1.0.0 )

 and I did
 USE=~doc ~alsa ~mysql ~gtk nopri nozaptel emerge asterisk

 do I need alsa, mysql, libpri or zaptel?

 tom
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RE: [Asterisk-Users] KSS/BLF on Asterisk

2004-10-24 Thread Jim Van Meggelen
[EMAIL PROTECTED] wrote:
 [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.

Thanks. I've taken a look at that beastie and I'll hack around with it
to see what it can do.


 Nick
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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Nicolás Gudiño
Hi Benj,

On Sun, 24 Oct 2004 23:39:06 +0900, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote:
 [snip URLs]
 
 yes, I have played with those and all I did achieve was making the
 recordings worse, but thanks anyway.
 
 However, it seems now that this is not a common problem so I have to
 check back with the customer whether these files are really the
 originals. You never know, they might have messed around with those.
 If anything else fails I will ask them to reorder the prompts once
 more and get them delivered as GSM.

I've never ordered from thevoice, but I have converted some MP3 to gsm
and after fighting with sox parameteres I came up with this:

sox in.wav -t gsm -r 8000 -g -b -c 1 out.gsm resample -ql vol 0.6

You can adjust the volume changing the last parameter. It works good
enough for me. With the wiki examples I got mixed result, like a
changed pitch, or the same pitch but a really slow speed for some
recordings.

Best regards,


-- 
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] Re: Direct SIP connection to Vonage service

2004-10-24 Thread Benjamin on Asterisk Mailing Lists
On Sun, 24 Oct 2004 15:27:53 +0200, Stewart Nelson [EMAIL PROTECTED] wrote:
 No, I don't want to replace existing gear.

fair enough.

 It would be expensive

that I don't agree with, especially not if you do it yourself, but anyway.

 There are other reasons, too.  For example, the Cisco 827-4V is very
 reliable, because it has no hard drive and no fans.  If *your*
 Asterisk system fails, you can zip over to Akihabara and get what
 you need, even on Sunday.  Rue Montgallet is not the same!

Unattended remote systems I run off Compact Flash instead of hard
drives. I can't beat the no fan thing though ;-)

 for me, it seems better to gradually migrate
 to a SIP-based system, with a single Asterisk server in Reno, and
 retaining present hardware.

I still think it would make sense to run one Asterisk server in Paris
and another in Reno. You don't really need more than that, but of
course it's your choice.

 When I'm in a hotel, stuck behind a NAT over which I have no control,
 sure, I'll use IAX to connect to the server (and tolerate the media
 proxy delays in that case.)

There are no media proxy delays if you use an IAX provider. IAX calls
skip intermediary nodes and go from end to end. That was the point of
my previous message.

If you use H.323 gateway with a SIP provider or vice versa, then you
will also have a media proxy in between because SIP gear cannot
reinvite H.323 gear, there's got to be something in between that
translates. If your entire route is all SIP, then you can go from end
to end. Like wise if youre entire route is all IAX, then you can go
from end to end, too.

 Have I missed something?

Yeah, wine and cheese is better, cheaper and more plentiful in Paris
than in Tokyo and that beats shopping in Akihabara anytime ;-)

rgds
benjk

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

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Re: [Asterisk-Users] random crash at startup

2004-10-24 Thread Tomas Carnecky
Steve Totaro wrote:
get the new versions of libpri zaptel and asterisk and install them in that
order.  should work.
I have libpri-1.0.0 and now I've reinstalled asterisk-1.0.0 but it still 
doesn't work, and there is no Digium hardware in the server so I don't 
need zaptel.

tom
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[Asterisk-Users] Failed to authenticate on INVITE to '601 ...

2004-10-24 Thread Ronald Wiplinger
I have installed the first time Asterisk,  (forgive me simple questions)
I have also installed the demo.
After testing demo (call 1000, call 600, ...) I changed in the 
extensions.conf:

; include = demo
include = incomingsipgate
include = sipgate.de
include = sipgate.col.uk
[incomingsipgate]
exten = 5552220,1,Dial(SIP/601,20,r)
exten = 4782156,1,Dial(SIP/602,20,r)
[sipgate.de]
exten = _0049X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten = _0049X.,2,Playback(invalid)
exten = _0049X.,3,Hangup
[sipgate.co.uk]
exten = _0044X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _0044X.,2,Playback(invalid)
exten = _0044X.,3,Hangup

in sip.conf I have:
register = 5552220:[EMAIL PROTECTED]/5552220
register = 4782156:[EMAIL PROTECTED]/4782156
[601]
type=friend
username=601
secret=pwd-601
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=601
nat=yes
caller-id=Ronald 1 601
[602]
type=friend
username=602
secret=pwd-602
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=602
nat=yes
caller-id=Ronald 2 602
[sipgate.de]
type=friend
username=5552220
secret=pwd-de
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.de
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=pwd-uk
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.co.uk
nat=yes
context=incomingsipgate
canreinvite=no


The console shows when I want to dial at sipgate.de  the number 1 
(test) or 5 (Voicemail):  00491

-- Executing Dial(SIP/601-ea8b, SIP/[EMAIL PROTECTED]|30|r) in new stack
-- Called [EMAIL PROTECTED]
Oct 24 23:00:48 NOTICE[213005]: c6779 handle_response: Failed to 
authenticate on INVITE to '601 sip:[EMAIL PROTECTED];tag=as25c3e254'
-- Nobody picked up in 3 ms
-- Executing Playback(SIP/601-ea8b, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Got SIP response 481 Call Leg Does Not Exist back from 217.10.79.9
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, 00491, 3) exited non-zero on 'SIP/601-ea8b'
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, h , 1) exited non-zero on 'SIP/601-ea8b'


What do I miss?
bye
Ronald
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
email;internet:[EMAIL PROTECTED]
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version:2.1
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Re: [Asterisk-Users] Fedora 2, Kudzu and X100P

2004-10-24 Thread Carlos Chavez
On Sun, 2004-10-24 at 05:10, Tzafrir Cohen wrote:

 One obvious solution is not to automatically load kudzu.
 
   chkconfig --remove kudzu
 
 Another obvious solution of the same sort is modprobing the zaptel
 module earlier in the boot process. 
 
 I can't seem to figure out , though, where kudzu takes its modue names
 from. I haven't bothred reading th source yet, though (not from
 /usr/share/hwdata, it seems)

The problem is that once Kudzu runs it configures linux to always load
the incorrect module for the card.  I have already erased kudzu from the
server, recompiled Zaptel and modified modules.dep by hand but if any
application runs a depmod -a the configuration for the other module
returns.  If I do a modprobe zaptel it will always load the other
module.

-- 
Carlos Chvez
Director de Tecnologa
Telecomunicaciones Abiertas de Mxico

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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Benjamin on Asterisk Mailing Lists
On Sun, 24 Oct 2004 12:01:19 -0300, Nicolás Gudiño [EMAIL PROTECTED] wrote:
 I've never ordered from thevoice, but I have converted some MP3

I guess you mean WAV

 to gsm
 and after fighting with sox parameteres I came up with this:
 
 sox in.wav -t gsm -r 8000 -g -b -c 1 out.gsm resample -ql vol 0.6

thanks a lot, I will try that.

rgds
benjk

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

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Re: [Asterisk-Users] New Channel Driver: chan_bluetooth

2004-10-24 Thread Martin List-Petersen
Yep, same here with Ericsson T610

Reason: AT+BRSF is not implemented in Ericsson Cellphones.

Kind regards,
Martin List-Petersen
http://www.marlow.dk/

On Wed, 2004-10-20 at 22:20, Jon Radon wrote:
 Running Asterisk CVS-HEAD-10/19/04-04:34:45, just tested with my Sony
 Ericsson T68i.  Couldn't get it to connect, got the error message
 below.  It then just sat saying it was negotiating.  I tried to
 disable BTP, but that didn't help.
 
 Oct 20 17:10:14 NOTICE[196620]: chan_bluetooth.c:1769 try_connect:
 Initialised bluetooth link to device t68i
 Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2152 handle_rd_data:
 Expected '\n'
 Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data:
 Expected '\r'
 Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data:
 Expected '\r'
 Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data:
 Expected '\r'
 Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data:
 Expected '\r'
 Oct 20 17:10:14 ERROR[196620]: chan_bluetooth.c:2083 handle_rd_data:
 Expected '\r'
 
 


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Re: [Asterisk-Users] random crash at startup

2004-10-24 Thread Steve Totaro
just try it.


- Original Message - 
From: Tomas Carnecky [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 11:05 AM
Subject: Re: [Asterisk-Users] random crash at startup


 Steve Totaro wrote:
  get the new versions of libpri zaptel and asterisk and install them in
that
  order.  should work.
 

 I have libpri-1.0.0 and now I've reinstalled asterisk-1.0.0 but it still
 doesn't work, and there is no Digium hardware in the server so I don't
 need zaptel.

 tom
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Re: [Asterisk-Users] Failed to authenticate on INVITE to '601 ... solved

2004-10-24 Thread Ronald Wiplinger
Ronald Wiplinger wrote:
I have installed the first time Asterisk,  (forgive me simple 
questions)

I have also installed the demo.

I solved it with the newest cvs version !!!
bye
Ronald

After testing demo (call 1000, call 600, ...) I changed in the 
extensions.conf:

; include = demo
include = incomingsipgate
include = sipgate.de
include = sipgate.col.uk
[incomingsipgate]
exten = 5552220,1,Dial(SIP/601,20,r)
exten = 4782156,1,Dial(SIP/602,20,r)
[sipgate.de]
exten = _0049X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten = _0049X.,2,Playback(invalid)
exten = _0049X.,3,Hangup
[sipgate.co.uk]
exten = _0044X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _0044X.,2,Playback(invalid)
exten = _0044X.,3,Hangup

in sip.conf I have:
register = 5552220:[EMAIL PROTECTED]/5552220
register = 4782156:[EMAIL PROTECTED]/4782156
[601]
type=friend
username=601
secret=pwd-601
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=601
nat=yes
caller-id=Ronald 1 601
[602]
type=friend
username=602
secret=pwd-602
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=602
nat=yes
caller-id=Ronald 2 602
[sipgate.de]
type=friend
username=5552220
secret=pwd-de
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.de
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=pwd-uk
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.co.uk
nat=yes
context=incomingsipgate
canreinvite=no


The console shows when I want to dial at sipgate.de  the number 1 
(test) or 5 (Voicemail):  00491

-- Executing Dial(SIP/601-ea8b, SIP/[EMAIL PROTECTED]|30|r) in new 
stack
-- Called [EMAIL PROTECTED]
Oct 24 23:00:48 NOTICE[213005]: c6779 handle_response: Failed to 
authenticate on INVITE to '601 sip:[EMAIL PROTECTED];tag=as25c3e254'
-- Nobody picked up in 3 ms
-- Executing Playback(SIP/601-ea8b, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Got SIP response 481 Call Leg Does Not Exist back from 217.10.79.9
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, 00491, 3) exited non-zero on 
'SIP/601-ea8b'
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, h , 1) exited non-zero on 'SIP/601-ea8b'


What do I miss?
bye
Ronald
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--
Ronald Wiplinger
Senior Software Engineer
AGP Telecom Co. Ltd.
Tel. (O) +886-2-2741-7890 # 7303, (M) +886-939-77-55-16
(from USA dial (408)253-3153 # 7303)
-Disclaimer---
This document is intended for transmission to the named recipient only. If you 
are not that person, you should note that legal rights reside in this 
document and you are not authorized to access, read, disclose, copy, use or 
otherwise deal with it and any such actions are prohibited and may be 
unlawful. The views expressed in this document are not necessarily those of 
AGP Telecom Co., Ltd. Notice is hereby given that no representation, contract 
or other binding obligation shall be created by this e-mail, which must be 
interpreted accordingly. Any representations, contractual rights or 
obligations shall be separately communicated in writing and signed in the 
original by a duly authorized officer of the relevant company. 
--

begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
email;internet:[EMAIL PROTECTED]
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version:2.1
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Re: [Asterisk-Users] Fedora 2, Kudzu and X100P

2004-10-24 Thread Steve Totaro
Just an idea, couldnt you remove the zaptel hardware, run kudzu and remove
the hardware module via kudzu then disable kudzu again?


- Original Message - 
From: Carlos Chavez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 11:15 AM
Subject: Re: [Asterisk-Users] Fedora 2, Kudzu and X100P


On Sun, 2004-10-24 at 05:10, Tzafrir Cohen wrote:

 One obvious solution is not to automatically load kudzu.

   chkconfig --remove kudzu

 Another obvious solution of the same sort is modprobing the zaptel
 module earlier in the boot process.

 I can't seem to figure out , though, where kudzu takes its modue names
 from. I haven't bothred reading th source yet, though (not from
 /usr/share/hwdata, it seems)

The problem is that once Kudzu runs it configures linux to always load
the incorrect module for the card.  I have already erased kudzu from the
server, recompiled Zaptel and modified modules.dep by hand but if any
application runs a depmod -a the configuration for the other module
returns.  If I do a modprobe zaptel it will always load the other
module.

-- 
Carlos Chvez
Director de Tecnologa
Telecomunicaciones Abiertas de Mxico

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Re: [Asterisk-Users] random crash at startup

2004-10-24 Thread Tomas Carnecky
Steve Totaro wrote:
just try it.
I installed the old 0.9 version of asterisk and now it works, even with 
libpri-1.0.0.

I've found out thet the kernel module ztdummy wasn't loaded while I 
tried to start asterisk, could this have been the problem?

tom
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Re: [Asterisk-Users] random crash at startup

2004-10-24 Thread Steve Totaro
more info on ztdummy and zaptel i am sure will solve your issue.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
- Original Message - 
From: Tomas Carnecky [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 11:42 AM
Subject: Re: [Asterisk-Users] random crash at startup


 Steve Totaro wrote:
  just try it.
 

 I installed the old 0.9 version of asterisk and now it works, even with
 libpri-1.0.0.

 I've found out thet the kernel module ztdummy wasn't loaded while I
 tried to start asterisk, could this have been the problem?

 tom
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[Asterisk-Users] Re: doublehash patch for 1.0.1

2004-10-24 Thread Randy Bush
 Just tried the patch you made with the latest CVS and it patches
 fine although it does not work.  Now when I hit # it does not
 send the DTMF to the other side at all.  Although hitting ##
 does get the transfer.  Now # doesn't do ANYTHING :)
 I'm not sure why that is, it works with all our phones
 (Grandstream BT101s and analog phones on Grandstream ATA286s).  I
 just tested by calling my bank's IVR.

applied patch.  went great.  now single # does not transfer and
double does.  but, i am having the same problem as matthew, the
# does not go through at dmtf.  all other keys go through as
dmtf, just not the #.  this is on a spa3k.

clearly * is receiving the #, as ## does do a transfer.  so why
is a single # not being sent onward as dtmf?

randy

---

ps. and i have a general wonder/question about this.  is someone
who uses a commercial pbx, say a meridian or whatever, unable to
use ivr systems because # is not sent?

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Re: [Asterisk-Users] Re: doublehash patch for 1.0.1

2004-10-24 Thread Matthew Marlowe
99% of the companies I call that say hit # after entering your
response, doesnt actually require the #.. I've only encountered a few
places that if you don't hit the # it ignores your response,
eventually sending you to an operator or hangs up on you.


On Sun, 24 Oct 2004 09:37:09 -0700, Randy Bush [EMAIL PROTECTED] wrote:
 
 
  Just tried the patch you made with the latest CVS and it patches
  fine although it does not work.  Now when I hit # it does not
  send the DTMF to the other side at all.  Although hitting ##
  does get the transfer.  Now # doesn't do ANYTHING :)
  I'm not sure why that is, it works with all our phones
  (Grandstream BT101s and analog phones on Grandstream ATA286s).  I
  just tested by calling my bank's IVR.
 
 applied patch.  went great.  now single # does not transfer and
 double does.  but, i am having the same problem as matthew, the
 # does not go through at dmtf.  all other keys go through as
 dmtf, just not the #.  this is on a spa3k.
 
 clearly * is receiving the #, as ## does do a transfer.  so why
 is a single # not being sent onward as dtmf?
 
 randy
 
 ---
 
 ps. and i have a general wonder/question about this.  is someone
 who uses a commercial pbx, say a meridian or whatever, unable to
 use ivr systems because # is not sent?
 
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-- 
MBM
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RE: [Asterisk-Users] Call Waiting

2004-10-24 Thread Henry Devito
If this is call waiting on the CO line, I found to flash the CO line you
have to (flash *0) to answer it.  If it is another station calling your
phone while you are on , a normal flash will do. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nikhil Jogia
Sent: Sunday, October 24, 2004 6:54 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call Waiting

Hi,

I have just set up an Asterisk box.it sure is a big job to get
everything perfect, especially when you have picky users.

Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
connected to the LAN.

1 of the lines connected to the X100P's goes straight to extension 1000
after a short greeting.

Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext
1000 hears a small beep every few seconds. This is obviously call
waiting.

My question is how do I answer that incoming call whilst on a call? I
have looked around, tried *0 and even 0*, the flash key, but to no avail
:(

If this is call waiting on the CO line, I found to flash the CO line you
have to (flash *0) to answer it.  If it is another station calling your
phone while you are on , a normal flash will do. 


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Re: [Asterisk-Users] IAXy setup

2004-10-24 Thread Andres Tello Abrego
Hummm.. thats cool..
I havent tougth about being re-provisioning the iaxy box :)...
But how do you detect the dns change? wich ddns company are u using?
Jim Van Meggelen wrote:

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Wilson Pickett
Sent: October 21, 2004 5:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXy setup

A little late, but about DNS for IAXY

request name resolution, no? Also, what happens if your IP address 
changes at the Asterisk end? You'll have to connect to all 
your IAXys 

to
Here's what I do: when the server ip changes, I auto 
reprovision the IAXy. It works better than the Grandstreams, 
which have to be rebooted when the ip changes. They do DNS, 
but only at boot time apparently! 
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Re: [Asterisk-Users] Call Waiting

2004-10-24 Thread Steve Totaro
ah good thinking, i didnt even factor CO call waiting into the equation


- Original Message - 
From: Henry Devito [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 1:19 PM
Subject: RE: [Asterisk-Users] Call Waiting


 If this is call waiting on the CO line, I found to flash the CO line you
 have to (flash *0) to answer it.  If it is another station calling your
 phone while you are on , a normal flash will do.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Nikhil Jogia
 Sent: Sunday, October 24, 2004 6:54 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Call Waiting

 Hi,
 
 I have just set up an Asterisk box.it sure is a big job to get
 everything perfect, especially when you have picky users.
 
 Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
 connected to the LAN.

 1 of the lines connected to the X100P's goes straight to extension 1000
 after a short greeting.

 Anyway, say ext 1000 is talking to ext 1001, and the line rings. Ext
 1000 hears a small beep every few seconds. This is obviously call
 waiting.

 My question is how do I answer that incoming call whilst on a call? I
 have looked around, tried *0 and even 0*, the flash key, but to no avail
 :(

 If this is call waiting on the CO line, I found to flash the CO line you
 have to (flash *0) to answer it.  If it is another station calling your
 phone while you are on , a normal flash will do.


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Re: [Asterisk-Users] Re: doublehash patch for 1.0.1

2004-10-24 Thread steve


 clearly * is receiving the #, as ## does do a transfer.  so why
 is a single # not being sent onward as dtmf?

I noticed on X-Lite that # in a dialstring is sent URL-encoded or similar, 
and Asterisk doesn't understand it.

Could this be something similar?  Perhaps sip debug will reveal?

Steve

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Re: [Asterisk-Users] Uniden UIP 200 Phone and NAT?

2004-10-24 Thread Ryan Courtnage
On Sat, 2004-23-10 at 19:43 -0500, Me wrote:
 Any chance you can pass me the Beta Version or let me know how to get it 
 myself?

I'm sorry, I can't distribute the beta.  You can ask Uniden support -
although I doubt they'd hand it out willingly.  I wouldn't expect much
more of a wait before an official version is released.

Ryan

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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread Kristian Kielhofner
Benjamin on Asterisk Mailing Lists wrote:
On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote:
http://www.theivrvoice.com/
would seem to imply otherwise.  I'd be a bit surprised if any company had
enough work to keep her employed full-time, so the works at Digium line
sounds a bit fishy to me.

I think when he wrote 'She does work at Digium' it was meant in the
sense of She is doing work at Digium', or 'She does (some) work for
Digium ;-)
rgds
benjk
I would lean towards she does some work for Digium.  Did you check out 
her webpage?  One, she lives in Canada, so she certainly does not work 
at Digium in the physical sense.  Two, her client list leads me to 
believe that Digium is probably one of her smaller clients.

Did you try to contact her directly?  She seems to imply on her site 
that customer satisfaction is pretty important to her.  Maybe she will 
fix them for you.  You did after all pay for these files, right? ;)

She must have a fairly good relationship with Digium, however, because 
she does have the official title of Asterisk Diva, and she was at 
Astricon.

--
Kristian Kielhofner
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Re: [Asterisk-Users] MusicOnHold() - how to restart player from the beginning on each call? (fwd)

2004-10-24 Thread Emilio Panighetti
Looks like what you want is not music on-hold, but rather a streaming 
server

On Oct 22, 2004, at 4:23 PM, Ryan Courtnage wrote:
On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote:
On Fri, 2004-10-22 at 05:56, Manfred Petz wrote:
[snip]
Is there a way to force MusicOnHold() to be restarted from the 
beginning for
each call which has been answered?
[snip]
Why?  What would be the point?
off the top of my head ... promotional messages.
Manfred - I don't think there is a graceful way to do this.  I know 
that
if you do a killall mpg123 at your command line, the next call MOH
answers will start playing the mp3s at the beginning.  Of course this
would affect others that are listening, but if you build out some logic
you might be able to make some use of it.

Ryan
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Re: [Asterisk-Users] MusicOnHold() - how to restart player from the beginning on each call? (fwd)

2004-10-24 Thread William Suffill
Why not just create a context that plays static msgs whenever someone
is transfered thereThank you for calling Monthly special etc
...
then transfer them back when the person at the biz picks up


On Sun, 24 Oct 2004 14:23:04 -0400, Emilio Panighetti [EMAIL PROTECTED] wrote:
 Looks like what you want is not music on-hold, but rather a streaming
 server
 
 
 
 On Oct 22, 2004, at 4:23 PM, Ryan Courtnage wrote:
 
  On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote:
  On Fri, 2004-10-22 at 05:56, Manfred Petz wrote:
  [snip]
  Is there a way to force MusicOnHold() to be restarted from the
  beginning for
  each call which has been answered?
  [snip]
 
  Why?  What would be the point?
 
  off the top of my head ... promotional messages.
 
  Manfred - I don't think there is a graceful way to do this.  I know
  that
  if you do a killall mpg123 at your command line, the next call MOH
  answers will start playing the mp3s at the beginning.  Of course this
  would affect others that are listening, but if you build out some logic
  you might be able to make some use of it.
 
  Ryan
 
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RE: [Asterisk-Users] Geotel integration with Asterisk

2004-10-24 Thread Greg Smith

Geotel is a company that Cisco bought which provides call control across
geographically dispersed locations.  The simplest application is being
able to query call queue status at another location.  For example, a
call comes in and can be sent to one of three call center locations.
Geotel can query each location to see who is the least busy for this
type of call.  Traditionally it has been VERY expensive.  

We provide some primitive Geotel functions in-the-cloud right now.  For
example, we can know how many live calls are going to a location before
we send the call.  We can set thresholds (e.g. if a location A has over
100 concurrent calls send them to location B).  Geotel can theoretically
provide this and carry it further.  I think there is some nice
enterprise reporting that can come from the Geotel as well.

G.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Saturday, October 23, 2004 4:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Geotel integration with Asterisk

Ok lets get this out of the way... WTF is Geotel?

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Greg Smith
 Sent: Saturday, October 23, 2004 4:19 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Geotel integration with Asterisk
 
 
 Has any one integrated to a Geotel with Asterisk?
 
 Thanks.
 
 Greg
 Advanta

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[Asterisk-Users] getting cid from spa3k pstn to *

2004-10-24 Thread Randy Bush
in order to get the cid from the spa3k to *, i need to turn on
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES

this produces a sip invite as follows:

Frame 1 (1092 bytes on wire, 1092 bytes captured)
Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 
(666.42.7.11)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
Session Initiation Protocol
Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-388cd0e7
From: CID Namesip:[EMAIL PROTECTED];tag=42d678b4c352ea69o1
To: sip:[EMAIL PROTECTED]
Remote-Party-ID: CID Namesip:[EMAIL 
PROTECTED];screen=yes;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061
Expires: 240
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 430
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Message body
Session Description Protocol

note that the From: has the cid, as does the Remote-Party-ID:.  and the
Contact: has the spa3k's id and display name.  as the sip.conf entry looks
like

[spa3k]
type=friend
host=dynamic
port=5061
auth=md5
secret=hidden
qualify=1000
dtmfmode=rfc2833
canreinvite=yes
context=spa3k-ext

the From: and/or Remote-Party-ID: cause asterisk respond with 407 Proxy
Authentication Required, to which the spa3k responds

Frame 3 (450 bytes on wire, 450 bytes captured)
Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 
(666.42.7.11)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
Session Initiation Protocol
Request-Line: ACK sip:[EMAIL PROTECTED] SIP/2.0
Method: ACK
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-388cd0e7
From: CID Namesip:[EMAIL PROTECTED];tag=42d678b4c352ea69o1
To: sip:[EMAIL PROTECTED];tag=as2741cf03
Call-ID: [EMAIL PROTECTED]
CSeq: 101 ACK
Max-Forwards: 70
Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 0

and it all goes to hell from there.

if i set the spa3k config to have
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO

Frame 1 (1072 bytes on wire, 1072 bytes captured)
Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11 
(666.42.7.11)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
Session Initiation Protocol
Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-f5998d8a
From: spa3k pstn sip:[EMAIL PROTECTED];tag=8fc58211a0dc60f2o1
To: sip:[EMAIL PROTECTED]
Remote-Party-ID: spa3k pstn sip:[EMAIL 
PROTECTED];screen=yes;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: spa3k pstn sip:[EMAIL PROTECTED]:5061
Expires: 240
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 430
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Message body
Session Description Protocol

the connection completes, but asterisk does not have the pstn caller id.

randy

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[Asterisk-Users] Re: [Asterisk-Dev] New Channel Driver: chan_bluetooth

2004-10-24 Thread Benjamin on Asterisk Mailing Lists
On Wed, 20 Oct 2004 13:37:28 +0100, Theo Zourzouvillys
[EMAIL PROTECTED] wrote:
 
 after a couple of days work banging my head against the wall (bloody standards
 my arse), i've got chan_bluetooth to a point where it's starting to function
 - certianly more than just proof of concept now.


Will this need a mobile phone with bluetooth support or can you use a
mere bluetooth headset as a client, if so how would you pick up and
hang up or dial?

I would really love to play with that, but as everybody knows, Japan
is s incredibly far advanced in mobile phones that the only phone
I could buy here which does have bluetooth works only as a data modem
for a Windoze notebook, no voice, no headset support, no addressbook
syncing, which is why I then bought an iPod for the $250 the phone
would have cost.

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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[Asterisk-Users] Asterisk Prepaid with MySQL

2004-10-24 Thread Nahuel Alejandro Ramos
Hi,
  Anyone could use Asterisk Prepaid with a MySQL database? Thanks.

   Nahuel Ramos.
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[Asterisk-Users] Compiling zaptel

2004-10-24 Thread Peer Oliver Schmidt
Hi,
I've been running * for a couple of month now. However, now i want to
run ztdummy. Compiling works (apart from some warning regarding
strict-aliasing), however installation gives missing Unresolved symbols:
if [ -d /etc/modutils ]; then \
/sbin/update-modules ; \
fi
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/wcusb.o
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/zaptel.o
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/ztd-eth.o
/sbin/depmod -a
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/wcusb.o
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/zaptel.o
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/ztd-eth.o
[ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample
/etc/zaptel.conf
I ran (as suggested on this list) depmod -ae with the following information:
pbx:/usr/src/zaptel# depmod -ae
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/wcusb.o
depmod: usb_submit_urb_R8c511495
depmod: usb_register_Re5a350f5
depmod: usb_deregister_R7002f0f3
depmod: usb_control_msg_R2ea78c08
depmod: usb_set_interface_Ra1eb4dc9
depmod: usb_set_configuration_Rd5d22e8d
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/zaptel.o
depmod: create_proc_entry_Rfe079653
depmod: remove_wait_queue_R9bd53081
depmod: proc_mkdir_Reb28b76c
depmod: remove_proc_entry_R63256fd5
depmod: __pollwait_R05dbf31f
depmod: add_wait_queue_R27b3fac5
depmod: register_chrdev_R007abe28
depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-686/misc/ztd-eth.o
depmod: skb_over_panic_R0f38fc6b
depmod: dev_queue_xmit_Rea2fb899
depmod: dev_remove_pack_Rdf0e2a5d
depmod: alloc_skb_R0f71f7ff
depmod: dev_get_by_name_R14968228
depmod: __kfree_skb_R72b0e92c
depmod: dev_add_pack_R52914681
depmod: skb_under_panic_Rf1c9f235
Here is a list of the relevant modules loaded:
pbx:/usr/src/zaptel# lsmod|grep usb
usb-uhci   23344   0  (unused)
usbcore62924   1  [usb-uhci]
pbx:/usr/src/zaptel# lsmod|grep ppp
ppp_generic20388   0  (unused)
slhc4784   0  [ppp_generic isdn]
Any and all help is greatly appreciated.
--
Best regards
Peer Oliver Schmidt
the internet company
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[Asterisk-Users] Reload cause Sound Volumn becomes very loud

2004-10-24 Thread R Wong
Hi all,
   I am running the Asterisk with CVS-HEAD-10/25/04.
   When I type reload in console, whatever the incoming/outgoing sound 
volumn becomes very loud until I stop the asterisk and restart it.
   It's running no problem before I've upgrade the asterisk. Is there any 
configuration I need to modify? Since I'm not able to find any information 
in this list nor voip-info.org (maybe I've overlook, if so please point me 
to the correct URL..)

   Thanks!
Regards,
R Wong 

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Re: [Asterisk-Users] Re: doublehash patch for 1.0.1

2004-10-24 Thread Lyle Giese
I don't know but it's IMHO, this should be just the opposite.  Single # for
a transfer and double ## to send the key on as DTMF.  How many objects in a
dialplan start with a #?

Lyle

- Original Message - 
From: Randy Bush [EMAIL PROTECTED]
To: Barton Hodges [EMAIL PROTECTED]
Cc: splatters [EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 11:37 AM
Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1


  Just tried the patch you made with the latest CVS and it patches
  fine although it does not work.  Now when I hit # it does not
  send the DTMF to the other side at all.  Although hitting ##
  does get the transfer.  Now # doesn't do ANYTHING :)
  I'm not sure why that is, it works with all our phones
  (Grandstream BT101s and analog phones on Grandstream ATA286s).  I
  just tested by calling my bank's IVR.

 applied patch.  went great.  now single # does not transfer and
 double does.  but, i am having the same problem as matthew, the
 # does not go through at dmtf.  all other keys go through as
 dmtf, just not the #.  this is on a spa3k.

 clearly * is receiving the #, as ## does do a transfer.  so why
 is a single # not being sent onward as dtmf?

 randy

 ---

 ps. and i have a general wonder/question about this.  is someone
 who uses a commercial pbx, say a meridian or whatever, unable to
 use ivr systems because # is not sent?

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RE: [Asterisk-Users] Re: doublehash patch for 1.0.1

2004-10-24 Thread Storm D. J. Petersen
What if you call an external system and get a voicemail. Press # to finish
your message .  you would have to press ##.

IMHO I think most users are not sophisticated enough to transfer calls.  If
they are they can press  ##.

Or am I missing something? :)

S.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese
Sent: Sunday, October 24, 2004 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1

I don't know but it's IMHO, this should be just the opposite.  Single # for
a transfer and double ## to send the key on as DTMF.  How many objects in a
dialplan start with a #?

Lyle

- Original Message -
From: Randy Bush [EMAIL PROTECTED]
To: Barton Hodges [EMAIL PROTECTED]
Cc: splatters [EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 11:37 AM
Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1


  Just tried the patch you made with the latest CVS and it patches
  fine although it does not work.  Now when I hit # it does not
  send the DTMF to the other side at all.  Although hitting ##
  does get the transfer.  Now # doesn't do ANYTHING :)
  I'm not sure why that is, it works with all our phones
  (Grandstream BT101s and analog phones on Grandstream ATA286s).  I
  just tested by calling my bank's IVR.

 applied patch.  went great.  now single # does not transfer and
 double does.  but, i am having the same problem as matthew, the
 # does not go through at dmtf.  all other keys go through as
 dmtf, just not the #.  this is on a spa3k.

 clearly * is receiving the #, as ## does do a transfer.  so why
 is a single # not being sent onward as dtmf?

 randy

 ---

 ps. and i have a general wonder/question about this.  is someone
 who uses a commercial pbx, say a meridian or whatever, unable to
 use ivr systems because # is not sent?

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Re: [Asterisk-Users] Re: doublehash patch for 1.0.1

2004-10-24 Thread Lyle Giese
I do some script type programing and have seen this in other uses.  IMHO, it
would be easier to program this way.  Single # go to transfer function.  Get
# as first character in transfer, send out the DTMF tones instead and drop
the request to transfer.

I could be all wet on this, but my feeble mind sezs this makes sense from a
programming perspective.

Lyle

- Original Message - 
From: Storm D. J. Petersen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 3:04 PM
Subject: RE: [Asterisk-Users] Re: doublehash patch for 1.0.1


 What if you call an external system and get a voicemail. Press # to
finish
 your message .  you would have to press ##.

 IMHO I think most users are not sophisticated enough to transfer calls.
If
 they are they can press  ##.

 Or am I missing something? :)

 S.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese
 Sent: Sunday, October 24, 2004 12:52 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1

 I don't know but it's IMHO, this should be just the opposite.  Single #
for
 a transfer and double ## to send the key on as DTMF.  How many objects in
a
 dialplan start with a #?

 Lyle

 - Original Message -
 From: Randy Bush [EMAIL PROTECTED]
 To: Barton Hodges [EMAIL PROTECTED]
 Cc: splatters [EMAIL PROTECTED]
 Sent: Sunday, October 24, 2004 11:37 AM
 Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1


   Just tried the patch you made with the latest CVS and it patches
   fine although it does not work.  Now when I hit # it does not
   send the DTMF to the other side at all.  Although hitting ##
   does get the transfer.  Now # doesn't do ANYTHING :)
   I'm not sure why that is, it works with all our phones
   (Grandstream BT101s and analog phones on Grandstream ATA286s).  I
   just tested by calling my bank's IVR.
 
  applied patch.  went great.  now single # does not transfer and
  double does.  but, i am having the same problem as matthew, the
  # does not go through at dmtf.  all other keys go through as
  dmtf, just not the #.  this is on a spa3k.
 
  clearly * is receiving the #, as ## does do a transfer.  so why
  is a single # not being sent onward as dtmf?
 
  randy
 
  ---
 
  ps. and i have a general wonder/question about this.  is someone
  who uses a commercial pbx, say a meridian or whatever, unable to
  use ivr systems because # is not sent?
 
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RE: [Asterisk-Users] Re: doublehash patch for 1.0.1

2004-10-24 Thread Storm D. J. Petersen
Personally I think like you... but I have to force myself to consider the
dim wits that use my PBX. :)  They are fat old men who barely understand
what a telephone is... let alone VOIP. :)

S.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese
Sent: Sunday, October 24, 2004 1:11 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1

I do some script type programing and have seen this in other uses.  IMHO, it
would be easier to program this way.  Single # go to transfer function.  Get
# as first character in transfer, send out the DTMF tones instead and drop
the request to transfer.

I could be all wet on this, but my feeble mind sezs this makes sense from a
programming perspective.

Lyle

- Original Message -
From: Storm D. J. Petersen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 3:04 PM
Subject: RE: [Asterisk-Users] Re: doublehash patch for 1.0.1


 What if you call an external system and get a voicemail. Press # to
finish
 your message .  you would have to press ##.

 IMHO I think most users are not sophisticated enough to transfer calls.
If
 they are they can press  ##.

 Or am I missing something? :)

 S.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese
 Sent: Sunday, October 24, 2004 12:52 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1

 I don't know but it's IMHO, this should be just the opposite.  Single #
for
 a transfer and double ## to send the key on as DTMF.  How many objects in
a
 dialplan start with a #?

 Lyle

 - Original Message -
 From: Randy Bush [EMAIL PROTECTED]
 To: Barton Hodges [EMAIL PROTECTED]
 Cc: splatters [EMAIL PROTECTED]
 Sent: Sunday, October 24, 2004 11:37 AM
 Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1


   Just tried the patch you made with the latest CVS and it patches
   fine although it does not work.  Now when I hit # it does not
   send the DTMF to the other side at all.  Although hitting ##
   does get the transfer.  Now # doesn't do ANYTHING :)
   I'm not sure why that is, it works with all our phones
   (Grandstream BT101s and analog phones on Grandstream ATA286s).  I
   just tested by calling my bank's IVR.
 
  applied patch.  went great.  now single # does not transfer and
  double does.  but, i am having the same problem as matthew, the
  # does not go through at dmtf.  all other keys go through as
  dmtf, just not the #.  this is on a spa3k.
 
  clearly * is receiving the #, as ## does do a transfer.  so why
  is a single # not being sent onward as dtmf?
 
  randy
 
  ---
 
  ps. and i have a general wonder/question about this.  is someone
  who uses a commercial pbx, say a meridian or whatever, unable to
  use ivr systems because # is not sent?
 
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[Asterisk-Users] Iaxy authentication

2004-10-24 Thread Me
Hello, working on trying to get the Iaxy setup from behind a NAT.
I have done everything the way I think it should be done but I can't seem to 
get dial tone and each time the device trys to register with * I get this 
message on the console:

***
Oct 24 15:15:11 NOTICE[131080]: chan_iax2.c:3865 register_verify: No 
registration for peer '100' (from MyIPwasHere)
***

Each time I actually pickup the phone connected to the Iaxy I get this:
**
Oct 24 15:16:55 NOTICE[131080]: chan_iax2.c:5390 socket_read: Rejected 
connect attempt from 67.166.254.124
**

Seems that the authentication is not working..
I have set the user: and pass: fields in the iaxy_ext.conf which is fed into 
the iaxy and set the accountcode: in iax.conf to be the same as the 
user: field in the file that is fed into the Iaxy, is that correct or does 
the username go somewhere else. I have set the secret: the same as the 
password: field in the file that's fed into the iaxy.

What have I missed here, I have dug and looked in the Wiki, Google, Digium 
site, list archives and there does not appear to be all that much 
documentation that I can find on setting this device up.

Any help would be appreciated..
Thanks,
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com 

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RE: [Asterisk-Users] KSS/BLF on Asterisk

2004-10-24 Thread Henry Devito


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

I've looked all over the wiki, and all the documentation I could get my
hands on, Where did you find anything about the hint priority?  I am
interested in trying to make this work.

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Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)

2004-10-24 Thread Me
Ok, all of this makes sense but I guess the bigger question is..
How does one check their voice mail and delete it by using a phone and 
dialing into *? Is there a magic extension or series of buttons to push to 
get someone into their mailbox?

Thanks,
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 7:22 AM
Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)


May be risky if your email is screwy but it solves your problem
Add:  delete=yes in your voicemail.conf.
- Original Message - 
From: Stephen R. Besch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, October 22, 2004 11:55 AM
Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue)


Todd Routhier - Lightwave Technologies, LLC. wrote:
 OK, this is a different flashing issue than the one that's being talked
 about.

 I have a few of these phones (GrandStream 101) and when a voicemail is
 received the light on the LED starts blinking and the dial tine
stutters,
 this is cool. BUT. How the  do I get it to stop, I have had 
 mine
 covered with paper for the last 2 days because the blinking LED panel 
 is
 driving me nuts. I have received the messages in my email and looked at
them
 on the web.

 I am thinking that I have to check them by calling into the Asterisk
system
 and mark them as read or something in order for this to quite.

 Two problems, I don't know how to check them by phone just yet and I
will
 likely never check them by phone.

You absolutely must get the message file deleted from the mail store. It
doesn't matter how you delete the message either. You can use the *
phone interface or simply delete the files associated with the message.
As soon as the files are gone from the INBOX, * sends a SIP command to
the phone to turn off the MWI. As long as there are any active messages
in the INBOX, the light stays on.
If you would like to be a guinea pig for my VB program that allows you
to manage your mail folders and messages from a Windows GUI, I'll send
you a copy. The only caveat is that I haven't yet found a way to get
perfect synchronization with file access to the mail store. The user
needs to be careful not to modify the INBOX while * is taking a new
message. For me this is usually not a problem, since I would never be
not answering the phone when I am listening to/moving/deleting messages
- as a result, * would never be writing a new message when I was using
the VB program.
The program requires the VB6 runtime (available from MS for free) and
that you have SAMBA running on the * server.
Sincerely,
Stephen R. Besch
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[Asterisk-Users] Unknown RTP codec 72 received

2004-10-24 Thread Danny Froberg
19 Question + this one and no answer;
Does anyone have a clue what causes Unknown RTP codec 72 received notice 
and how to fix it?

Regards
Danny
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Re: [Asterisk-Users] Re: [Asterisk-Dev] New Channel Driver: chan_bluetooth

2004-10-24 Thread Martin List-Petersen
On Sun, 2004-10-24 at 20:08, Benjamin on Asterisk Mailing Lists wrote:
 On Wed, 20 Oct 2004 13:37:28 +0100, Theo Zourzouvillys
 [EMAIL PROTECTED] wrote:
  
  after a couple of days work banging my head against the wall (bloody standards
  my arse), i've got chan_bluetooth to a point where it's starting to function
  - certianly more than just proof of concept now.
 
 Will this need a mobile phone with bluetooth support or can you use a
 mere bluetooth headset as a client, if so how would you pick up and
 hang up or dial?

It supports both.

Regarding the headset, i have not seen how that works yet, but i would
say you would need to enter the number somewhere (maybe special prefix
on any phone + phoneno. to get the call to the headset)

For the cellphone, this is really grand stuff:
I originially was hoping for using the cellphone as a handset to the
Asterisk box, but it's quite the oposite.

Theo has basically obsoleted the expensive PSTN to GSM converters.
Asterisk connects via bluetooth to the phone and uses it as a FXO line.
Neat stuff.

I'm seeing forward to, how this work progresses, because it's really
promissing.

Kind regards,
Martin List-Petersen


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Re: [Asterisk-Users] Re: [Asterisk-Dev] New Channel Driver: chan_bluetooth

2004-10-24 Thread Benjamin on Asterisk Mailing Lists
On Sun, 24 Oct 2004 21:42:49 +0100, Martin List-Petersen
[EMAIL PROTECTED] wrote:
 Regarding the headset, i have not seen how that works yet, but i would
 say you would need to enter the number somewhere (maybe special prefix
 on any phone + phoneno. to get the call to the headset)

Oh well, I guess I could use the manager API to initiate the call from
my Powerbook. Cool! So, where do I get a Bluetooth headset now? I
don't think they sell those over here either.

 Asterisk connects via bluetooth to the phone and uses it as a FXO line.
 Neat stuff.

Too bad. Japan and Korea are the only two countries on the planet
without GSM. Even Afghanistan has got GSM now. We don't even get a
proper Bluetooth keitai. Frustrating.

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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RE: [Asterisk-Users] G.729 . . . I SMELL SMOKE!

2004-10-24 Thread Steven Critchfield
On Sat, 2004-10-23 at 01:06 -0400, Jim Van Meggelen wrote:
 Few will disagree that the careful application of netiquette will be a
 benefit to any newsgroup/mailing list/board; and top posting is
 something that should be used sparingly. Nevertheless, top posting is
 not the horrid crime some might have us believe. When used
 appropriately, it serves very well, and only causes offense to the
 ideologues. Me too-type top posing is usually of no benefit, but when
 someone is commenting on a tangled and involved thread, it can make
 sense to frame the entirety of the thread in a thoughtful top post.

Don't forget the same people who refuse to trim the bottom of the post
and we end up with 20(your case only 1) copy of the mailing list footer.

 Then we get to the most dangerous beast, the abusive, expert troll. This
 is someone who clearly is very intelligent and articulate, and could
 argue their value due to a) their willingness to contribute, b) their
 level of knowledge and c) their fantastic writing skills. Unfortunately,
 these folks reduce their value to almost nothing by virtue of their
 pathetic lack of any manners whatsoever. They will drive away more
 people than they help -- and that doesn't bother them in the slightest.
 What a waste of talent.

As I am sure to be painted by the above brush, let me offer just a small
point here. I have had just a bit of time to think this over after
politely listenening to the same argument from another person this
weekend. 

You seem to not realize that those who are knowlegable are only so due
to the vast amount of time we put into learning. I'm sure there are many
people who are like me and are trying to spend a lot of time learning
several projects that have no overlap. While we seek all this knowlege,
I hope the others like me actually try and do things outside of the
computer world as well. 

Now I want you to realize that many of the really newbie or lazy (these
are NOT equal in the level I detest) questions that are answerable with
a quick browse of the wiki or a simple google search end up being
equivalent to SPAM in my mailbox as I try and search for information
that furthers my knowlege. Understand that I learn from looking at what
others are doing, and answers to others questions. 

So when you try and run off those who know a fair amount but don't meet
your manners requirement, I want you to think about why you feel newbie
or lazy users should be of higher value than those with the knowlege?
Why do you wish to preserve their participation at the detriment to
those who have more answers than questions?

Hoping that the person I talked to this weekend is actually reading. I
don't consider myself any more important than anyone else in this list,
but rather I like others, wish to defend this channel of information
from descending below useful signal to noise ratio.

 We all understand that it is generally best to avoid feeding the trolls,
 but every now and then the townsfolk must grab shovels and pitchforks,
 and drive these beasts back into the caves from whence they came; where,
 one hopes, they will contemplate the value of a few simple manners, and
 perhaps even one day to attempt to give to the community without the
 needless rancour.

-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)

2004-10-24 Thread Steve Totaro
better get to reading.
Basically you need to create an extension that points to voicemailmain.


- Original Message - 
From: Me [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 4:30 PM
Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)


 Ok, all of this makes sense but I guess the bigger question is..

 How does one check their voice mail and delete it by using a phone and
 dialing into *? Is there a magic extension or series of buttons to push to
 get someone into their mailbox?

 Thanks,
  Todd
 --
 Start Your Own ISP!
 http://www.YourOwnISP.com

 - Original Message - 
 From: Steve Totaro [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
 Sent: Sunday, October 24, 2004 7:22 AM
 Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)


  May be risky if your email is screwy but it solves your problem
 
  Add:  delete=yes in your voicemail.conf.
 
 
  - Original Message - 
  From: Stephen R. Besch [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  [EMAIL PROTECTED]
  Sent: Friday, October 22, 2004 11:55 AM
  Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue)
 
 
  Todd Routhier - Lightwave Technologies, LLC. wrote:
   OK, this is a different flashing issue than the one that's being
talked
   about.
  
   I have a few of these phones (GrandStream 101) and when a voicemail
is
   received the light on the LED starts blinking and the dial tine
  stutters,
   this is cool. BUT. How the  do I get it to stop, I have had
   mine
   covered with paper for the last 2 days because the blinking LED panel
   is
   driving me nuts. I have received the messages in my email and looked
at
  them
   on the web.
  
   I am thinking that I have to check them by calling into the Asterisk
  system
   and mark them as read or something in order for this to quite.
  
   Two problems, I don't know how to check them by phone just yet and I
  will
   likely never check them by phone.
  
  You absolutely must get the message file deleted from the mail store.
It
  doesn't matter how you delete the message either. You can use the *
  phone interface or simply delete the files associated with the message.
  As soon as the files are gone from the INBOX, * sends a SIP command to
  the phone to turn off the MWI. As long as there are any active messages
  in the INBOX, the light stays on.
 
  If you would like to be a guinea pig for my VB program that allows you
  to manage your mail folders and messages from a Windows GUI, I'll send
  you a copy. The only caveat is that I haven't yet found a way to get
  perfect synchronization with file access to the mail store. The user
  needs to be careful not to modify the INBOX while * is taking a new
  message. For me this is usually not a problem, since I would never be
  not answering the phone when I am listening to/moving/deleting messages
  - as a result, * would never be writing a new message when I was using
  the VB program.
 
  The program requires the VB6 runtime (available from MS for free) and
  that you have SAMBA running on the * server.
 
  Sincerely,
 
  Stephen R. Besch
 
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RE: [Asterisk-Users] Unknown RTP codec 72 received

2004-10-24 Thread Robert Jackson


 -Original Message-
 From: Danny Froberg [mailto:[EMAIL PROTECTED] 
 Sent: Sunday, October 24, 2004 4:37 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Unknown RTP codec 72 received
 
 
 19 Question + this one and no answer;
 
 Does anyone have a clue what causes Unknown RTP codec 72 
 received notice 
 and how to fix it?
 
 Regards
 Danny
 

I receive this message when I call from X-Lite.  I notice that
it is usually when I am sending DTMF digits.  I could be using 
the wrong dtmfmode (using info), but I am not sure.

This message is rather annoying so I would definitely like to 
see if anyone else has gotten it figured out.

Robert Jackson
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Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)

2004-10-24 Thread Steve Totaro
http://www.voip-info.org/tiki-searchresults.php?words=voicemailwhere=pages
- Original Message - 
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 5:27 PM
Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)


 better get to reading.
 Basically you need to create an extension that points to voicemailmain.


 - Original Message - 
 From: Me [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
 Sent: Sunday, October 24, 2004 4:30 PM
 Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)


  Ok, all of this makes sense but I guess the bigger question is..
 
  How does one check their voice mail and delete it by using a phone and
  dialing into *? Is there a magic extension or series of buttons to push
to
  get someone into their mailbox?
 
  Thanks,
   Todd
  --
  Start Your Own ISP!
  http://www.YourOwnISP.com
 
  - Original Message - 
  From: Steve Totaro [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  [EMAIL PROTECTED]
  Sent: Sunday, October 24, 2004 7:22 AM
  Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)
 
 
   May be risky if your email is screwy but it solves your problem
  
   Add:  delete=yes in your voicemail.conf.
  
  
   - Original Message - 
   From: Stephen R. Besch [EMAIL PROTECTED]
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   [EMAIL PROTECTED]
   Sent: Friday, October 22, 2004 11:55 AM
   Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue)
  
  
   Todd Routhier - Lightwave Technologies, LLC. wrote:
OK, this is a different flashing issue than the one that's being
 talked
about.
   
I have a few of these phones (GrandStream 101) and when a voicemail
 is
received the light on the LED starts blinking and the dial tine
   stutters,
this is cool. BUT. How the  do I get it to stop, I have had
mine
covered with paper for the last 2 days because the blinking LED
panel
is
driving me nuts. I have received the messages in my email and
looked
 at
   them
on the web.
   
I am thinking that I have to check them by calling into the
Asterisk
   system
and mark them as read or something in order for this to quite.
   
Two problems, I don't know how to check them by phone just yet and
I
   will
likely never check them by phone.
   
   You absolutely must get the message file deleted from the mail store.
 It
   doesn't matter how you delete the message either. You can use the *
   phone interface or simply delete the files associated with the
message.
   As soon as the files are gone from the INBOX, * sends a SIP command
to
   the phone to turn off the MWI. As long as there are any active
messages
   in the INBOX, the light stays on.
  
   If you would like to be a guinea pig for my VB program that allows
you
   to manage your mail folders and messages from a Windows GUI, I'll
send
   you a copy. The only caveat is that I haven't yet found a way to get
   perfect synchronization with file access to the mail store. The user
   needs to be careful not to modify the INBOX while * is taking a new
   message. For me this is usually not a problem, since I would never be
   not answering the phone when I am listening to/moving/deleting
messages
   - as a result, * would never be writing a new message when I was
using
   the VB program.
  
   The program requires the VB6 runtime (available from MS for free) and
   that you have SAMBA running on the * server.
  
   Sincerely,
  
   Stephen R. Besch
  
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RE: [Asterisk-Users] Geotel integration with Asterisk

2004-10-24 Thread dean collins
From what I read about a year ago was that it was a carrier hosted
solution that actually controlled the ss7 switching at the exchange
(basically no call costs from tromboning, and was only implemented into
an ip-centrex or hosted call centre application.

Are you saying that enterprises can buy something similar and control
the carriers switching?



Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Smith
Sent: Sunday, October 24, 2004 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Geotel integration with Asterisk


Geotel is a company that Cisco bought which provides call control across
geographically dispersed locations.  The simplest application is being
able to query call queue status at another location.  For example, a
call comes in and can be sent to one of three call center locations.
Geotel can query each location to see who is the least busy for this
type of call.  Traditionally it has been VERY expensive.  

We provide some primitive Geotel functions in-the-cloud right now.  For
example, we can know how many live calls are going to a location before
we send the call.  We can set thresholds (e.g. if a location A has over
100 concurrent calls send them to location B).  Geotel can theoretically
provide this and carry it further.  I think there is some nice
enterprise reporting that can come from the Geotel as well.

G.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Saturday, October 23, 2004 4:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Geotel integration with Asterisk

Ok lets get this out of the way... WTF is Geotel?

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Greg Smith
 Sent: Saturday, October 23, 2004 4:19 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Geotel integration with Asterisk
 
 
 Has any one integrated to a Geotel with Asterisk?
 
 Thanks.
 
 Greg
 Advanta

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RE: [Asterisk-Users] Iaxy authentication

2004-10-24 Thread Joe Dennick
I bought out IAXy devices from NetXUSA, who also sent me this short
installation document which I've copied below:

Quick Start Guide for Digium IAXY Device


   Determine the IP address of each unit by viewing the logs on
your DHCP Server to see which IP address your IAXY has taken,  you
cannot reserve IP address based on the MAC address. You may however
define a static IP address after the initial setup.
   Login to Asterisks CVS and checkout  iaxyprov, then do a make
   Once your in the iaxyprov directory edit the iaxy.sample.conf
file or create a new configuration file
   Define the IP address or if you wish to continue to use DHCP,
assign a unique user name for your device, password, codec, and the
Asterisk Server it will register with. For an example see the
iaxy.sample.conf file
   Once you have defined your iaxy.conf file proceed to run
iaxyprov. The format is as follows:
./iaxyprov IP Address iaxy.conf File name

You should receive a confirmation back that the data was
received, Repeat these same steps for each  
IAXY device, changing only the user name and IP address if not
using DHCP
   Next, you need to configure the Iax.conf file under
/etc/asterisk. Create each user, see the sample configuration below

[100]   ;User 100
type=friend
username=100
secret=1234
context=default
disallow=all
allow=ulaw
mailbox=1234

   Lastly, the format for the IAXY devices in extensions.conf is
IAX2/User
Exten = 100,1,Dial(IAX2/100)


For further questions please contact NETXUSA Technical support Monday 
Friday 9 am est  6 pm est or refer to the Asterisk user list for help.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Me
Sent: Sunday, October 24, 2004 3:24 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Iaxy authentication


Hello, working on trying to get the Iaxy setup from behind a NAT.

I have done everything the way I think it should be done but I can't
seem to 
get dial tone and each time the device trys to register with * I get
this 
message on the console:

***
Oct 24 15:15:11 NOTICE[131080]: chan_iax2.c:3865 register_verify: No 
registration for peer '100' (from MyIPwasHere)
***

Each time I actually pickup the phone connected to the Iaxy I get this:

**
Oct 24 15:16:55 NOTICE[131080]: chan_iax2.c:5390 socket_read: Rejected 
connect attempt from 67.166.254.124
**

Seems that the authentication is not working..

I have set the user: and pass: fields in the iaxy_ext.conf which is fed
into 
the iaxy and set the accountcode: in iax.conf to be the same as the 
user: field in the file that is fed into the Iaxy, is that correct or
does 
the username go somewhere else. I have set the secret: the same as the

password: field in the file that's fed into the iaxy.

What have I missed here, I have dug and looked in the Wiki, Google,
Digium 
site, list archives and there does not appear to be all that much 
documentation that I can find on setting this device up.

Any help would be appreciated..

Thanks,
Todd

--
Start Your Own ISP!
http://www.YourOwnISP.com 

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Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)

2004-10-24 Thread Me
Thanks Steve, it's not that I have not been reading (ask my wife how many 
nights I have slept in the last week), and it's not that there is not a huge 
amount of info out there. The problem I am having is finding the info I need 
in any sort of organized way.

The searches I do sometimes come up with solutions but otherwise I am left 
to browse the Wiki which has tons of info, it's just not very obvious how to 
find it most of the time.

Thanks for the link, I am sure this will help!
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message - 
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 4:37 PM
Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)


http://www.voip-info.org/tiki-searchresults.php?words=voicemailwhere=pages
- Original Message - 
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 5:27 PM
Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)


better get to reading.
Basically you need to create an extension that points to voicemailmain.
- Original Message - 
From: Me [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 4:30 PM
Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different issue)

 Ok, all of this makes sense but I guess the bigger question is..

 How does one check their voice mail and delete it by using a phone and
 dialing into *? Is there a magic extension or series of buttons to push
to
 get someone into their mailbox?

 Thanks,
  Todd
 --
 Start Your Own ISP!
 http://www.YourOwnISP.com

 - Original Message - 
 From: Steve Totaro [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
 Sent: Sunday, October 24, 2004 7:22 AM
 Subject: Re: [Asterisk-Users] Re: Grandstream Flashing (different 
 issue)


  May be risky if your email is screwy but it solves your problem
 
  Add:  delete=yes in your voicemail.conf.
 
 
  - Original Message - 
  From: Stephen R. Besch [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  [EMAIL PROTECTED]
  Sent: Friday, October 22, 2004 11:55 AM
  Subject: [Asterisk-Users] Re: Grandstream Flashing (different issue)
 
 
  Todd Routhier - Lightwave Technologies, LLC. wrote:
   OK, this is a different flashing issue than the one that's being
talked
   about.
  
   I have a few of these phones (GrandStream 101) and when a 
   voicemail
is
   received the light on the LED starts blinking and the dial tine
  stutters,
   this is cool. BUT. How the  do I get it to stop, I have 
   had
   mine
   covered with paper for the last 2 days because the blinking LED
panel
   is
   driving me nuts. I have received the messages in my email and
looked
at
  them
   on the web.
  
   I am thinking that I have to check them by calling into the
Asterisk
  system
   and mark them as read or something in order for this to quite.
  
   Two problems, I don't know how to check them by phone just yet and
I
  will
   likely never check them by phone.
  
  You absolutely must get the message file deleted from the mail 
  store.
It
  doesn't matter how you delete the message either. You can use the *
  phone interface or simply delete the files associated with the
message.
  As soon as the files are gone from the INBOX, * sends a SIP command
to
  the phone to turn off the MWI. As long as there are any active
messages
  in the INBOX, the light stays on.
 
  If you would like to be a guinea pig for my VB program that allows
you
  to manage your mail folders and messages from a Windows GUI, I'll
send
  you a copy. The only caveat is that I haven't yet found a way to get
  perfect synchronization with file access to the mail store. The user
  needs to be careful not to modify the INBOX while * is taking a new
  message. For me this is usually not a problem, since I would never 
  be
  not answering the phone when I am listening to/moving/deleting
messages
  - as a result, * would never be writing a new message when I was
using
  the VB program.
 
  The program requires the VB6 runtime (available from MS for free) 
  and
  that you have SAMBA running on the * server.
 
  Sincerely,
 
  Stephen R. Besch
 
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RE: [Asterisk-Users] KSS/BLF on Asterisk

2004-10-24 Thread Jim Van Meggelen
[EMAIL PROTECTED] wrote:
 Get a hint! :-)
 
 Check out the hint priority in extensions.conf.  There are also
 some details in the wiki.
 
 I've looked all over the wiki, and all the documentation I
 could get my hands on, Where did you find anything about the
 hint priority?  I am interested in trying to make this work.

The only references I could find to it were here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20exten
sions
here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
and here:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg49781.htm
l

Not much to work from, but the lack of documentation on this feature is
probably signifigant. I realize that this is going to be something that
requires more research on my part, as no one appears to be using it very
much.


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Re: [Asterisk-Users] MusicOnHold() - how to restart player from the beginning on each call? (fwd)

2004-10-24 Thread Matthew Marlowe
What Manfred wants to do is not that uncommon.  I've used the method
that William has suggested in the past.  On a lot of corporate phone
systems this is a simple option in the programming.  Another way is to
simply advertise your specials over your music on hold and repeat
them... Hoping that the person hears the special.. The only reason
they wouldn't is if the customer was on hold for too small of an
amount of time, which the customer will appreciate more. :)


On Sun, 24 Oct 2004 14:26:50 -0400, William Suffill
[EMAIL PROTECTED] wrote:
 Why not just create a context that plays static msgs whenever someone
 is transfered thereThank you for calling Monthly special etc
 ...
 then transfer them back when the person at the biz picks up
 
 
 
 
 On Sun, 24 Oct 2004 14:23:04 -0400, Emilio Panighetti [EMAIL PROTECTED] wrote:
  Looks like what you want is not music on-hold, but rather a streaming
  server
 
 
 
  On Oct 22, 2004, at 4:23 PM, Ryan Courtnage wrote:
 
   On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote:
   On Fri, 2004-10-22 at 05:56, Manfred Petz wrote:
   [snip]
   Is there a way to force MusicOnHold() to be restarted from the
   beginning for
   each call which has been answered?
   [snip]
  
   Why?  What would be the point?
  
   off the top of my head ... promotional messages.
  
   Manfred - I don't think there is a graceful way to do this.  I know
   that
   if you do a killall mpg123 at your command line, the next call MOH
   answers will start playing the mp3s at the beginning.  Of course this
   would affect others that are listening, but if you build out some logic
   you might be able to make some use of it.
  
   Ryan
  
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-- 
MBM
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[Asterisk-Users] Connection to a H323 system

2004-10-24 Thread Ronald Wiplinger
I found in Google a h323.conf file, but not on my Asterisk installation.
Do I need to do more than h323.conf ???
I have a h323 phone and would like to replace it as one connection to my 
Asterisk, 

Thanks for your hints.
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] ACT Gateways

2004-10-24 Thread Joseph
Has anybody tested any gateways from ACT:
http://www.act-tel.com.tw/Index2.htm

They have four different configurations:
4xFXS - 4xFXO
2xFXS - 2xFXO
1xFXS - 1xFXO
4xFXS

I emailed them but they didn't bother the respond.

-- 
#Joseph
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Re: [Asterisk-Users] Connection to a H323 system

2004-10-24 Thread Steve Totaro
vi /usr/src/asterisk/channels/h323/h323.conf.sample

vi /usr/src/asterisk/channels/h323/README


- Original Message - 
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 8:01 PM
Subject: [Asterisk-Users] Connection to a H323 system


 I found in Google a h323.conf file, but not on my Asterisk installation.

 Do I need to do more than h323.conf ???

 I have a h323 phone and would like to replace it as one connection to my
 Asterisk, 

 Thanks for your hints.

 bye

 Ronald







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Re: [Asterisk-Users] iLBC/PCM16 Huge Cost

2004-10-24 Thread Trevor Peirce
Okay, I have removed the IDE Controller and am now using onboard.  The 
problems below still exist--

Trevor Peirce wrote:
show translation still reveals the iLBC column in the 8700 to 9600 
range though.  LPC10's row is also in the 900s.

show translation recalc 10 still causes the * console to stop 
responding as well.

When I first bootup *, it consumes nearly 100% of CPU, all in user... 
system is less than 1%.  The time that it monopolizes the processor 
varies as well.

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Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible

2004-10-24 Thread David Boyd

On Sun, 2004-10-24 at 10:24, Steve Totaro wrote:
 I know she works at Digium but they probably go down the street to a real
 sound stage to do the recordings via 3rd party.
 
 A sound stage is a facility used to create and process professional
 recordings.  They can be used by anyone employed by an company.
 
SNIP..

Take a look at the Astricon links, I believe that she is in Canada and
works for several groups.  



Dave



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RE: [Asterisk-Users] Geotel integration with Asterisk

2004-10-24 Thread David Boyd
On Sun, 2004-10-24 at 17:52, dean collins wrote:
 From what I read about a year ago was that it was a carrier hosted
 solution that actually controlled the ss7 switching at the exchange
 (basically no call costs from tromboning, and was only implemented into
 an ip-centrex or hosted call centre application.
 
 Are you saying that enterprises can buy something similar and control
 the carriers switching?
 
 
 
 Cheers,
 Dean
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Greg Smith
 Sent: Sunday, October 24, 2004 2:33 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Geotel integration with Asterisk
 
 
 Geotel is a company that Cisco bought which provides call control across
 geographically dispersed locations.  The simplest application is being
 able to query call queue status at another location.  For example, a
 call comes in and can be sent to one of three call center locations.
 Geotel can query each location to see who is the least busy for this
 type of call.  Traditionally it has been VERY expensive.  
 
 We provide some primitive Geotel functions in-the-cloud right now.  For
 example, we can know how many live calls are going to a location before
 we send the call.  We can set thresholds (e.g. if a location A has over
 100 concurrent calls send them to location B).  Geotel can theoretically
 provide this and carry it further.  I think there is some nice
 enterprise reporting that can come from the Geotel as well.
 
 G.
SNIP...


The GEOtel solution uses a type of interface that was originally
designed for  tie-in to the MCI network. The MCI network uses something
call a DAP(data access point) the DAP performs a database lookup anytime
that an 800,888,866,877 or virtual network number is dialed on their
network. This lookup is done via SS7 and returns the appropriate routing
information ie.. Switch and trunk group with appropriate DNIS to the
originating switch which then routes the call to the proper termination 
location.  The GEOtel solution actually works like a wedge into the call
routing info. By using an adjunct processor that is in contact with the
customers network switches/ACDs the DAP actually queries the adjunct
processor for the proper routing data, and returns the appropriate info
for call termination.  The return data is based on whatever rules that
the adjunct uses for the call lookup.  The original trial for this
service was used by MCI corporate for their own Customer service network
Galaxy class ACD's made by Rockwell.  The adjunct would poll the ACD's
and determine queuing times as well as time of day number of operators
etc, and return proper routing information.  This was called Intelligent
Routing Service (IRS) but the marketing group decided that Intelligent
Call Routing was a better name.  


Hope this was informative in some way :)

Dave  

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RE: [Asterisk-Users] KSS/BLF on Asterisk

2004-10-24 Thread Henry Devito
I am buying a Snom phone this week.  I will play with this feature and see
what I can get going.  I will share my findings.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Van
Meggelen
Sent: Sunday, October 24, 2004 6:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] KSS/BLF on Asterisk

[EMAIL PROTECTED] wrote:
 Get a hint! :-)
 
 Check out the hint priority in extensions.conf.  There are also
 some details in the wiki.
 
 I've looked all over the wiki, and all the documentation I
 could get my hands on, Where did you find anything about the
 hint priority?  I am interested in trying to make this work.

The only references I could find to it were here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20exten
sions
here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
and here:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg49781.htm
l

Not much to work from, but the lack of documentation on this feature is
probably signifigant. I realize that this is going to be something that
requires more research on my part, as no one appears to be using it very
much.


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[Asterisk-Users] Howto get voicemail $VM_ vars into externnotify script?

2004-10-24 Thread Patrick
Hi all,

I am trying to slap together a script that will email2sms the details of
the voicemails left on my * box to my gsm phone. I can't figure out how
to get my script to pick up the voicemail vars like ${VM_MSGNUM},
${VM_DATE}, ${VM_MAILBOX}, ${VM_CALLERID}, ${VM_DUR}. Right now I have
this:

#!/bin/sh
/bin/mail -s Voicemail received with details:
Msg nr  : `echo $VM_MSGNUM`
Msg date: `echo $VM_DATE`
For : `echo $VM_MAILBOX`
From: `echo $VM_CALLERID`
Length  : `echo $VM_DUR` [EMAIL PROTECTED]

Which doesn't work as all the $VM_ vars show up blank. 

Any ideas?

TIA,
Patrick

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[Asterisk-Users] Several FXS Ports

2004-10-24 Thread James Dumais
hello list.

looking for a way to have several FXS ports on an asterisk box, lets say
oh... 300, just for shoots and giggles. would i need special telco
equipment? if so, what kind? i already have a 23 inch cabinet, which i'm
told telco equipment uses 23 inch. any insight on this would be greatly
appreciated.

Thanks in advance
-- 
James W Dumais
ABSS::Networks
http://www.abss.ca/
1(705)725-9124 / 1(800)473-2121

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Re: [Asterisk-Users] Several FXS Ports

2004-10-24 Thread Steven Critchfield
On Sun, 2004-10-24 at 22:02 -0400, James Dumais wrote:
 hello list.
 
 looking for a way to have several FXS ports on an asterisk box, lets say
 oh... 300, just for shoots and giggles. would i need special telco
 equipment? if so, what kind? i already have a 23 inch cabinet, which i'm
 told telco equipment uses 23 inch. any insight on this would be greatly
 appreciated.

Do you realize this EXACT question came up as recent as this month on
this EXACT list.

http://lists.digium.com/pipermail/asterisk-users/2004-October/065518.html
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] G.729 . . . I SMELL SMOKE!

2004-10-24 Thread Mike Boger Jr

- Original Message - 
From: Steven Critchfield [EMAIL PROTECTED]
 You seem to not realize that those who are knowlegable are only so due
 to the vast amount of time we put into learning. I'm sure there are many
 people who are like me and are trying to spend a lot of time learning
 several projects that have no overlap. While we seek all this knowlege,
 I hope the others like me actually try and do things outside of the
 computer world as well.

I agree with you. I also spend signifigant time in learning things about
Asterisk and other projects.

 Now I want you to realize that many of the really newbie or lazy (these
 are NOT equal in the level I detest) questions that are answerable with
 a quick browse of the wiki or a simple google search end up being
 equivalent to SPAM in my mailbox as I try and search for information
 that furthers my knowlege. Understand that I learn from looking at what
 others are doing, and answers to others questions.

SPAM is in a way not just uneccesary solicitation but debate as well. The
kind of debate that nobody wins. the kind where the participants agree to
disagree. These postings take a toll on a list such as this. (and now i'm
guilty of it too) But my point is that this list generates a huge amount of
mail (and noise) I would prefer the former over the latter.

 So when you try and run off those who know a fair amount but don't meet
 your manners requirement, I want you to think about why you feel newbie
 or lazy users should be of higher value than those with the knowlege?
 Why do you wish to preserve their participation at the detriment to
 those who have more answers than questions?

I think that everyone who has participated in or used Asterisk can help
someone else or learn from someone else. It doesn't make the helper smarter
than the one asking the question. It allows someone who has been down a
particular path to help guide someone down that same path. It's really all
about community.

 Hoping that the person I talked to this weekend is actually reading. I
 don't consider myself any more important than anyone else in this list,
 but rather I like others, wish to defend this channel of information
 from descending below useful signal to noise ratio.

Again I agree. We are a community and a community is made up of differing
views, opinions, and beliefs. It's through a common interest we are all
together and therefore we must insure that the very things that bring us
together, that is the quest for knowledge about Asterisk and helping others
with our experiences with Asterisk are preserved.

  We all understand that it is generally best to avoid feeding the trolls,
  but every now and then the townsfolk must grab shovels and pitchforks,
  and drive these beasts back into the caves from whence they came; where,
  one hopes, they will contemplate the value of a few simple manners, and
  perhaps even one day to attempt to give to the community without the
  needless rancour.

I wish to apologize to Kevin Walsh. While I did not write the above, or
agree with Kevin, It was my sentiment.
at the time. Kevin I know you are an interested member of this community and
you have contributed your knowledge in a
meaningful way in the past. I hope you can forgive my rudeness.

Best Regards,

Mike Boger Jr.


 -- 
 Steven Critchfield [EMAIL PROTECTED]

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