Curt Shaffer wrote:
I am experiencing a situation where I am getting intermittent choppy
audio. Here is the network layout:
Termination provider -> IAX2 over the Internet -> 20Mb fiber
connection -> router -> Asterisk
My ATA connection goes into the router between the fiber and the
Asteris
I am experiencing a situation where I am getting intermittent choppy audio.
Here is the network layout:
Termination provider -> IAX2 over the Internet -> 20Mb fiber connection ->
router -> Asterisk
My ATA connection goes into the router between the fiber and the Asterisk
server on another i
Dovid B wrote:
ROTFL. The US patent system is treated with contempt in Hong Kong? You
have no idea how EXTREME legislation in Hong Kong against IP 'theft'
is in Hong Kong.
I find this hard to believe since most hack attempts to my box's
originate from IP's in China.
Welcome to China. M
On Sat, 2007-04-07 at 23:52 -0500, Eric "ManxPower" Wieling wrote:
> The device doing the IP/TDM conversion should be the device that sets
> the gains correctly. The same applies to echo canceling.
As I stated, this started with the warning of Novice Question :-).
Eric, can you elaborate on
Hello Armin (and happy easter),
thanks for you continuing support.
> Can you please try HEAD version of SVN trunk (443)?
Did checkout the 443.
It works without any verbosity.
THANK YOU! I'll buy you a beer, if you ever happen to come to the
northern part of Germany.
--
Best regards
Peer Oliv
On Tue, 3 Apr 2007, Armin Schindler wrote:
> On Tue, 3 Apr 2007, Peer Oliver Schmidt wrote:
> > Hello Armin,
> >
> > thanks a lot for your help.
> >
> > > Can you please do the same with 'showcapimsgs=2'?
> > > It may give more info on the commands itself, maybe some parameters are
> > > wrong he
Yuan LIU wrote:
>> From: "Arun Kumar" <[EMAIL PROTECTED]>
>> Date: Sun, 8 Apr 2007 05:25:58 -0700
>>
>> Hi,
>>
>> In my dial plan I've configured two trunks to make outbound calls (trunk1
>> and trunk2) to same service provider but I want when any of my exten
>> starts
>> with _2. should goto trunk
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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asterisk-users mailing list
To U
From: "Arun Kumar" <[EMAIL PROTECTED]>
Date: Sun, 8 Apr 2007 05:25:58 -0700
Hi,
In my dial plan I've configured two trunks to make outbound calls (trunk1
and trunk2) to same service provider but I want when any of my exten starts
with _2. should goto trunk2 and there should be some kind of distu
I am assuming this:
Call comes in, the Dial happens and for whatever reason the destination
cannot be reached. You then want to play a message to the caller.
Just put the "g" option on the end of Dial and then check the
HANGUPCAUSE. The destination has already hungup, but the caller has not
you can easily increment the buffer size changing include/asterisk/manager.h
#define AST_MAX_MANHEADER_LEN 256
chage that line for something like this
#define AST_MAX_MANHEADER_LEN 512
and recompile Asterisk,
Is the only way I know,
Regards
On 4/8/07, Thomas Winter <[EMAIL PROTECTED]> wrote
Once the call is hung up it is too late. I need to interpret the SIP
response codes prior to hangup so I can play an appropriate recorded voice
announcement.
On 4/9/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Eric Bishop wrote:
> Hi all,
>
> I want to implement certain actions based
Eric Bishop wrote:
Hi all,
I want to implement certain actions based on SIP response codes. Is there a
similar variable such as ${DIALSTATUS} that comes back with the relevant
SIP
response code for a call?
I believe there is SIPGetHeader, but Asterisk tries to translate
whatever code it get
Hi all,
I want to implement certain actions based on SIP response codes. Is there a
similar variable such as ${DIALSTATUS} that comes back with the relevant SIP
response code for a call?
--- Thanks
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Hi,
I use Originate to make a call.
I have problems to bring my vars into the channel.
Are there restrictions more then only 24 vars at mentioned at
www.voip-info.org?
Any workaround to get this running?
WARNING[4641]: manager.c:1365 get_input: Dumping long line with no return from
127.0.0.1
Hi,
In my dial plan I've configured two trunks to make outbound calls (trunk1
and trunk2) to same service provider but I want when any of my exten starts
with _2. should goto trunk2 and there should be some kind of disturbance
(like some noise or some background noise) when my calls goes to tru
Steve Underwood wrote:
> Dovid B wrote:
>
>>
>>
>>> ROTFL. The US patent system is treated with contempt in Hong Kong?
>>> You have no idea how EXTREME legislation in Hong Kong against IP
>>> 'theft' is in Hong Kong.
>>
>>
>>
>> I find this hard to believe since most hack attempts to my box's
>>
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To U
And if they get you black-listed you can always signup with Verizon...
On 4/8/07, Dean Collins <[EMAIL PROTECTED]> wrote:
There's no way for them to tell if you have asterisk on the fxo port BUT
they will terminate your account if you hook it up as the outbound for an
office pumping call after
Dovid B wrote:
ROTFL. The US patent system is treated with contempt in Hong Kong?
You have no idea how EXTREME legislation in Hong Kong against IP
'theft' is in Hong Kong.
I find this hard to believe since most hack attempts to my box's
originate from IP's in China.
What exactly would atta
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