On Mon, Apr 27, 2009 at 07:37:05AM +0100, --[ UxBoD ]-- wrote:
> Hi,
>
> Built a new server at the weekend and install Asterisk 1.6.0.9 and
> IAX and SIP work great :) The one problem I am having is getting the
> OpenVox (TDM400 type card) to work. It is successfully identified
> using WCTDM k
Hi,
Built a new server at the weekend and install Asterisk 1.6.0.9 and IAX and SIP
work great :) The one problem I am having is getting the OpenVox (TDM400 type
card) to work. It is successfully identified using WCTDM kernel module and
dahdi_scan picks it up just fine. The issue is when I try
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
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
ho
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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On Sun, Apr 26, 2009 at 1:28 PM, jonas kellens wrote:
> part of extensions.conf:
>
> *exten => 11,1,Answer()*
> *exten => 11,n,NoOp(CallerID : ${CALLERID(all)})*
> *exten => 11,n,Playback(/tmp/welkom-tcs.alaw)*
> *exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)*
> *; wordt doorgeroutee
On Mon, 27 Apr 2009 14:37:23 David Backeberg wrote:
> On Sun, Apr 26, 2009 at 8:33 AM, Michael wrote:
> > Is it possible to force T38 for all invocations ReceiveFAX() ?
>
> If it's not T.38, it should instead be audio over G.711 or similar
> codec. Are your faxes going through as audio? If not, th
On Sun, Apr 26, 2009 at 8:33 AM, Michael wrote:
> Is it possible to force T38 for all invocations ReceiveFAX() ?
If it's not T.38, it should instead be audio over G.711 or similar
codec. Are your faxes going through as audio? If not, that's a strong
indication that you have a SIP configuration is
On Mon, 27 Apr 2009 04:31:14 you wrote:
> Michael wrote:
> > Anyway this is a great example of why MICROSOFT is worth billions, and
> > Linux has to be given away. Not because Microsoft is L33T but because the
> > majority of the stuff sold for Windows works out of the box.
>
> For what it's worth,
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer '3516533812' is now Reachable. (98ms / 2000ms)
[Apr 26 12:08:49] WARNING[32273]: app_dial.c:1
Michael wrote:
> On Mon, 27 Apr 2009 03:40:12 you wrote:
>
>> Slightly off topic, but M$ 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
>> t
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
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 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
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
fromdomai
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 yo
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'
[2009-04-26
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]
type=frie
part of extensions.conf:
exten => 11,1,Answer()
exten => 11,n,NoOp(CallerID : ${CALLERID(all)})
exten => 11,n,Playback(/tmp/welkom-tcs.alaw)
exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)
; wordt doorgerouteerd naar context open, maar indien gesloten :
exten => 11,n,NoOp(Oproep tijdens
Michael wrote:
> On Mon, 27 Apr 2009 03:40:12 you wrote:
>
>> Slightly off topic, but M$ 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
>> t
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
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
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
a
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-inter
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
An
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 -
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-b
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
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
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 wrote:
> Sending works but on receive it keeps failing - reporting back 'tra
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.
_
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 t
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
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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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:
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
> sie
Ok cheers.
Any idea when 1.6 goes stable for prod?
- Original Message
From: Mike
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, 24 April, 2009 0:54:59
Subject: Re: [asterisk-users] Parked calls for multiple customers
No, but as I understand it 1.6 would
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