Ok cheers.
Any idea when 1.6 goes stable for prod?
- Original Message
From: Mike l...@virtutel.ca
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 24 April, 2009 0:54:59
Subject: Re: [asterisk-users] Parked calls for multiple
On Fri, Apr 24, 2009 at 12:09:50PM +0200, Wolfgang Pichler wrote:
hi all,
we do have some troubles with zaptel timing source - we have a setup
with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk
does some handling - calls are leaving on digium card 1 - going to a
siemens
Vincent wrote:
www.voip-info.org/wiki/view/Asterisk+embedded+systems
Thanks Steve. I knew about this list, but I wanted to make sure there
weren't other, more complete sources about the subject.
So at this point, it seems like it boils down to this:
Soekris
PCEngines
Atcom (IP01:
carl Lougher wrote:
Ok cheers.
Any idea when 1.6 goes stable for prod?
Theoretically it already has, however as was the case with 1.4, I
suggest you tread very carefully when it comes to migrating to 1.6.
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Hello,
It's happens around 40 calls and above …
The **machine** accepts number of invites(we can see by tcpdump ) , but
asterisk sees part of them (we can see by CLI log) , and when it does ,
asterisk accepting an invite it reply the initator. (as it should ) – but
the rest of invites are
On 26 Apr 2009, at 09:17, Paul Chambers wrote:
Vincent wrote:
www.voip-info.org/wiki/view/Asterisk+embedded+systems
Thanks Steve. I knew about this list, but I wanted to make sure there
weren't other, more complete sources about the subject.
So at this point, it seems like it boils down to
Is it possible to force T38 for all invocations ReceiveFAX() ?
Receiving fax always worked OK on Callweaver though I could put
SipT38Switchover() into the dial plan.
I can't with Digium fax, and it always fails at the point it decides to switch
to T38.
Turn on fax detection so that echo training will not attempt to run on
the fax handshake...
Andrew lathama Latham
TuxTone Inc.
http://TuxTone.com
andrew.lat...@tuxtone.com
On Sat, Apr 25, 2009 at 10:21 PM, Michael as...@nettrust.co.nz wrote:
Sending works but on receive it keeps failing -
On Mon, 27 Apr 2009 01:36:25 you wrote:
Turn on fax detection so that echo training will not attempt to run on
the fax handshake...
Andrew lathama Latham
TuxTone Inc.
http://TuxTone.com
andrew.lat...@tuxtone.com
Thanks.
How is this done on a T38 fax channel please?
PS: I see no mention
On Mon, 27 Apr 2009 02:02:12 you wrote:
When configured with echo training (default) Asterisk attempts to
train its self on all calls. You have to tell it to either not train
on all calls or turn on fax detection.
Andrew lathama Latham
How do I turn this on please? I can't find any mention
1.6.1 svn 190575:
CC=cc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent menuselect
make[1]: Entering directory
`/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect'
gcc -m64 -march=native -mtune=native -floop-interchange
-floop-strip-mine -floop-block -c -o
On Sun, Apr 26, 2009 at 10:26:12AM -0400, sean darcy wrote:
1.6.1 svn 190575:
CC=cc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent menuselect
make[1]: Entering directory
`/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect'
gcc -m64 -march=native
On Mon, 27 Apr 2009 02:33:31 you wrote:
have a look at the dahdi or zaptel configs and search for echo In
the last 9+ years of working with * I have never had a manual, whats
it look like..?
Andrew lathama Latham
TuxTone Inc.
http://TuxTone.com
andrew.lat...@tuxtone.com
And I also
sean darcy schrieb:
1.6.1 svn 190575:
CC=cc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent menuselect
make[1]: Entering directory
`/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect'
gcc -m64 -march=native -mtune=native -floop-interchange
Slightly off topic, but M$ is worth billions because they started in 1976 or
so, became the de facto standard, and were pretty cutthroat in the way they
do business. They have a profit motive and have always taken the path that
makes them bigger with bigger profits, even to the point of fighting
On Mon, 27 Apr 2009 03:40:12 you wrote:
Slightly off topic, but M$ is worth billions because they started in 1976
or so, became the de facto standard, and were pretty cutthroat in the way
they do business. They have a profit motive and have always taken the path
that makes them bigger with
Michael wrote:
Anyway this is a great example of why MICROSOFT is worth billions, and Linux
has to be given away. Not because Microsoft is L33T but because the majority
of the stuff sold for Windows works out of the box.
For what it's worth, their fax software doesn't work very well out of
Michael wrote:
On Mon, 27 Apr 2009 03:40:12 you wrote:
Slightly off topic, but M$ is worth billions because they started in 1976
or so, became the de facto standard, and were pretty cutthroat in the way
they do business. They have a profit motive and have always taken the path
that makes
part of extensions.conf:
exten = 11,1,Answer()
exten = 11,n,NoOp(CallerID : ${CALLERID(all)})
exten = 11,n,Playback(/tmp/welkom-tcs.alaw)
exten = 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)
; wordt doorgerouteerd naar context open, maar indien gesloten :
exten = 11,n,NoOp(Oproep tijdens
Hi,
have some problem with incoming calls from sipgate. This was working in
1.4 but in 1.6 I get a 401 Unauthorized :-(.
Sipgate has mentioned that I have to change the type to friend, but it
is already friend, so what's wrong?
Kind regards,
Michael
Here is the sip.conf:
[sipgate_out]
With 1.6.1 svn:
[2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout:
-- Registration for '17470121...@proxy01.sipphone.com' timed out, trying
again (Attempt #30)
[2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable
to lookup 'proxy01.sipphone.com'
Probably the phone is using a wrong DNS entry.
Try changing the Proxy address to the IP address and see if the phone
registers.
If so, work backwards from there.
Change DNS setting to some other DNS. And, remember you have to set an IP
DNS, you can't use a URL to look up a DNS. (Assuming you
Hi,
looks like I've found the solution by myself. The sipgate_out context
needs the parameter
insecure=invite
also I missed to set the context for the dialplan.
So in sip.conf using
--
[sipgate_out]
type=friend
context=extern
insecure=invite
nat=yes
username=1234567
fromuser=1234567
On 25/04/2009 4:29 p.m., Sam Hawkin wrote:
Hi,
Thanks for your reply
I have tried the HUMAN as you suggested , but still my problem does not
get solved.
I am answering the call and then running the amd.
Below is the log.
Few things.
1. Put an answer before the AMD line.
2. Put a
On 25/04/2009 1:55 a.m., Marco Sambo wrote:
Hi all,
I try to install FOP. It's very nice.
In documentation I red that from my dial plan I can launch a popup
window with UserEvent() application.
I try to follow FOP documentation but I can't popup anything. My
structure is:
- server 1:
On 24/04/2009 2:22 p.m., Saurabh Nirkhey wrote:
I have written an asterisk manager client which creates an outbound
call using Asterisk manager API's Originate action.
when the call is connected I run 3 applications on it.
1)read a dtmf digit from user
2)A customized application which I have
Michael wrote:
On Mon, 27 Apr 2009 03:40:12 you wrote:
Slightly off topic, but M$ is worth billions because they started in 1976
or so, became the de facto standard, and were pretty cutthroat in the way
they do business. They have a profit motive and have always taken the path
that makes
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer '3516533812' is now Reachable. (98ms / 2000ms)
[Apr 26 12:08:49] WARNING[32273]:
On Mon, 27 Apr 2009 04:31:14 you wrote:
Michael wrote:
Anyway this is a great example of why MICROSOFT is worth billions, and
Linux has to be given away. Not because Microsoft is L33T but because the
majority of the stuff sold for Windows works out of the box.
For what it's worth, their
On Sun, Apr 26, 2009 at 8:33 AM, Michael as...@nettrust.co.nz wrote:
Is it possible to force T38 for all invocations ReceiveFAX() ?
If it's not T.38, it should instead be audio over G.711 or similar
codec. Are your faxes going through as audio? If not, that's a strong
indication that you have a
On Mon, 27 Apr 2009 14:37:23 David Backeberg wrote:
On Sun, Apr 26, 2009 at 8:33 AM, Michael as...@nettrust.co.nz wrote:
Is it possible to force T38 for all invocations ReceiveFAX() ?
If it's not T.38, it should instead be audio over G.711 or similar
codec. Are your faxes going through as
On Sun, Apr 26, 2009 at 1:28 PM, jonas kellens jonas.kell...@telenet.bewrote:
part of extensions.conf:
*exten = 11,1,Answer()*
*exten = 11,n,NoOp(CallerID : ${CALLERID(all)})*
*exten = 11,n,Playback(/tmp/welkom-tcs.alaw)*
*exten = 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)*
*;
I am looking for Video Conference Software (Open Source) , But but not for
free Trial..
please give reference about it.
Thanks
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As what you said, it is very difficult to control if meetme is created
for each call. Playing a message after party A answers if a choice
but party A will still need to hear ring after the message. She may
still feel weird.
Just want to know the purpose of parameter async. Can anyone tell me
Hi,
Thanks for your reply.
I have tried as you suggested, I does not even come upto NoOp()
It hangups after AMD.
I have decreased the silence threshold from 256 to 100 and 50.
below is the log.
-- Executing Answer(SIP/sip-38ea, ) in new stack
-- Executing AMD(SIP/sip-38ea, ) in new stack
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