Displaying 20 results from an estimated 5000 matches similar to: "still no solution for me, if one provider fails."
2006 Apr 10
1
RE: still no solution for me, if one provider
>Our user places a call, the gateway responds with no sound at all, or
>hangs up, or gives busy tone.
>
>How can we get to the next provider?
>
>I have now:
>exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-a)
>;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-b)
>;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-c)
>exten =>
2006 Apr 16
1
[Fwd: Re: voicemail email-from]
Ronald Wiplinger wrote:
> Steve Totaro wrote:
>> Ronald Wiplinger wrote:
>>> kevin ling wrote:
>>>> Hi,
>>>>
>>>> Check the vm_general.inc file
>>>>
>>>>
>>> Where should this file be?
>>>
>>>
>>> bye
>>>
>>> Ronald Wiplinger
>>>
>>>
>> You
2005 Jul 17
2
DNS SRV
I have added in my zone file;
_sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com.
As I understand it should mean that any sip connection to
<anyname>@elmit.com should go to the udp port 5060 at the host
vpb.elmit.com.
In Asterisk's extensions.conf I have in the context [default]
exten => ronald,1,Dial(${PHONE_615},60,tr)
exten => ronald,2,Voicemail,u615@office
exten =>
2006 Jan 27
2
Name/username (sip show peers)
How can I make it more readable?
Name/username
601/601
123456789/123456789
voipbuster/abcd
601 = hotline
123456789 = Peter Pan
only voipbuster/abcd is easy read/understandable!
bye
Ronald Wiplinger
2006 Jun 04
3
transfer & other features
*CLI> show features
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # ##
Attended Transfer *2
One Touch Monitor *1
Disconnect Call * *0
Dial option is tTwWr
I tried to call from 601 to 615
601 keys in *0
2006 Apr 04
1
voipstunt: "Forbidden - wrong password ..."
voipstunt: "Forbidden - wrong password on authentication for INVITE to ...."
I have paid, the password was not changed, ... I have no idea why.
Is there anything what I can do to get this "failed" call over to
another provider, so that the user can complete the call?
(Dialstatus was an idea, but the line does not show up in CLI)
[Apr 5 09:22:36] -- Executing
2006 Apr 26
6
I am looking for a webphone on MY SITE
I am looking for a way of not to install a softphone, preferable as a
link on a web site to a webphone on MY SITE !!!
Has anybody an idea for that? AJAX?
bye
Ronald Wiplinger
2005 May 17
4
Is SKYPE a threat or should we do something (together)
Skype is very succesfsfull and get more and more users, ... we can
ignore them, accept them or do something,...
My suggestion is that we try to do something, ...
If we would peer to each other, than we get soon also a great amount of
users together, and than our service becomes more valuable, ...
Let's discuss advantages and disadvantages!
bye
Ronald
--
Ronald Wiplinger (CEO of
2007 Feb 14
5
Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the
bandwidth to different users. Each user should get e.g., 512kB/s plus
256kB/s dedicated for VoIP.
What kind of device can I use for that ? (managing switch ??? which one?)
bye
Ronald Wiplinger
2006 Mar 31
4
How to check if a phone / line is used?
In the past I used SetGroup and CheckGroup to figure out if my allowed
providers lines are all used or not.
Since most of my provider have given me a single line anyway, I wonder
if there is a way to check if this (provider) line is taken already.
How can I do that?
Same is with the phone. How can I see in CLI if a phone is now in use or
not?
"Sip show peers" shows me just if it is
2005 Jul 28
12
Can you caculate with me?
before I accuse somebody to "overbill" I would like you to calculate
with me:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for the answers: 0.7269 or 0.2929 ???
bye
Ronald Wiplinger
2006 Jun 24
5
ASTCC: How to reset periodically all "card in use" flag back?
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.
I have created a number the user can dial to reset this flag. However,
that is written in the manual!!! Who reads a manual anyway!!!!
I want to make to reset all in use flag with a program. Has anybody done
it, or has a better idea?
My idea
2005 Jul 23
2
ASTCC gives me only the time, but no cost
I try to track down an error that causes that Astcc just reports the time, but not the costs.
I could narrow the problem down into this sub routine:
sub calccost() {
my ($adjconn, $adjcost, $answeredtime, $increment) = @_;
eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment) * $increment };
my $cost;
print STDERR "Adjusted time is $adjtime, cost is $adjcost with
2007 Feb 14
2
moving WiFi phone
Can anybody tell me how I can set-up multiple access points with
overlapping coverage, so that a moving WiFi phone user can continuesly
use the phone.
bye
Ronald Wiplinger
2005 Jul 23
1
astcc timestamps
The time stamps in ASTCC are useless as they are now:
Fri Jul 22 15:06:25 2005
Wouldn't it be better to use something like:
2005-07-22 15:06:24 Fri
I want to sort the records by date, but with the format now it is
impossible... or do I miss something?
bye
Ronald Wiplinger
2005 Aug 11
2
list in asterisk cli is getting too long
How can I use something like |more in CLI ?
The lists are getting too long, like sip show users
bye
Ronald Wiplinger
2005 Sep 11
1
first character in line 11 missing
I would like to know if somebody else experienced that:
sip show peers will always drop the first character of the 11th line.
while sip show peers like [0-9,a-z] will not drop any character.
Can anybody test this, please?
bye
Ronald Wiplinger
2006 Apr 02
1
Who is on a call?
I would like to know which extension number is engaged in a call.
show channels shows me:
*CLI> show channels
Channel Location State
Application(Data)
SIP/asterisk.elmit.com-0 690@default:2 Up
Echo()
SIP/8807-066 690@newcontext Up Echo()
2 active channels
2 active calls
but it is not
2006 Apr 12
1
free video (soft) phone available?
I am using eyebeam and I am happy with it. However, it is boring just to
talk to my son in the other room.
Whenever I try to convince somebody to buy eyebeam, they are scared of
the price.
Is there a free video soft phone available, that will work with eyebeam
/ asterisk?
bye
Ronald Wiplinger
2006 Jun 24
2
Is anybody using XEN in conjunction with Asterisk and/or Openser?
Is anybody using XEN in conjunction with Asterisk and/or Openser?
I would like to get some info about such an environment and experience
reports.
bye
Ronald Wiplinger