Hey guys I just got a message from 4psa saying that they have a beta up
of their new software, which is basically asterisk, with a solid
interface. Anyway I just thought I would share the info:
http://forum.4psa.com/showthread.php?t=455
Take it for a ride around the block and tell them what y
Matthew Warren wrote:
Yes it is an addon of Plesk, thats stating the obvious.
I'm really confused how you think this is an addon of plesk? Sure it
looks like plesk, and feels like plesk, but what idiot would run a voip
system on top of a shared hosting server especially one as insecure as
I'm having a real problem with one of my linksys pap2. On outgoing
calls the callee will ring, but caller (pap2) will not here it ring
When the callee answers, no audio is transmitted either way. Asterisk
reports the call connected and bridged correctly.
Now the kicker is that sometimes it
nnect to a stun server, or DMZ your ATA, or port
forward all needed ports to the ATA's internal IP.
FYI, on several applications where I set port forwarding... I needed to set
nat=no to get it to work.
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf O
e it for a ton of
customers, I'd suggest building one so you can manage it yourself... of
course if you have money to burn, you could invest in a good SBC.
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miles Scruggs
Sent: Thursday, May 2
Using sip connections some peers are not able to transmit or recieve
audio. All peers are setup the same aside from the NAT settings. The
call will go through, called device will ring, but when it answers there
is no audio connection. From the callee, they will not here the rings,
only silen
or a short number.
Essentially a keepalive for any routers in the middle. If you have
multiple phones behind a remote NAT, make sure they are using
different ports.
Miles Scruggs wrote:
Using sip connections some peers are not able to transmit or recieve
audio. All peers are setup the same
l a good idea to use qualify=yes in your asterisk (sip.conf)
for each extension since it keeps port mappings open and active on
your linksys. Otherwise your Linksys port mapping may expire and an
incoming call will be seen as unsolicited traffic and block it.
Thanks,
Steve Totaro
Miles Scruggs
s are reflected in
asterisk.
Miles Scruggs wrote:
Well I just set the port to 5061, and no other devices on this end
have that port. I still have the same problems though. The strange
thing is that I have better luck calling the asterisk box itself
rather than an outside line, but even that
s there" is a little hard to work
with.
Is this a double NAT or is your asterisk box on a routable IP? If it
is double NAT, forget it.
Thanks,
Steve
Miles Scruggs wrote:
yup everything is there:
Name/username HostDyn Nat ACL Port
Status pap2-2/pap2-2
Derek Whitten wrote:
Miles Scruggs wrote:
Hmm all your questions are covered in this email, but I'll summarize it
again in this reply:
Server: 1.2.7.1 direct connection to the Internet
config settings:
[pap2]
type=friend
secret=something
qualify=yes
nat=yes
host=dynamic
canreinvi
Steve Totaro wrote:
Miles Scruggs wrote:
Derek Whitten wrote:
Miles Scruggs wrote:
Hmm all your questions are covered in this email, but I'll
summarize it
again in this reply:
Server: 1.2.7.1 direct connection to the Internet
config settings:
[pap2]
type=friend
secret=something
qu
I want to setup a fragment of my dialplan to dial an ext at the same
time as another, but only if the other is avalible for instance this
sudo code
var $ext1
var $ext2
var $ext3
if ($ext1) {
dial($ext1&$ext3)
}elseif($ext2) {
dial($ext2&$ext3)
}else{
dial($ext3)
}
Anyone know how I
I'm really confused on how to use these two options together:
A while back:
JustRumours
edited this page:
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
and added a little section about dynamic conferences. the 'e' option is
repeated all over the page as the savior of dynamic conferences
ut needing an AGI.
Look here
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
bp
On 6/16/06, *Miles Scruggs* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
I want to setup a fragment of my dialplan to dial an ext at the same
time as another, but
ound calling, and perform the logic based on the feedback.
I know I'm well outside the realm of the standard "please enter the ext
of the person you are calling" /voicemail standard systems, but I'm
guessing this is exactly
If anyone could please help me with this, that would be very helpful.
Miles Scruggs wrote:
Just starting to enjoy the full features of asterisk, I do have a
couple questions though, that I can't seem to find answers for in the
wiki, just wondering if someone could light my way.
af
Hey,
I would like in the course of dial plan logic, to trigger a separate
outbound call. If that outbound call is answered, and if that certain
key response is detected then it will bridge the incoming call to the
newly dialed outbound call.
What I want to accomplish is that when a caller d
I would also like to know how to do this, it really defeats the whole
purpose of the list if you reply off list.
Please post that to the list.
Miles
Shaun wrote:
Sent you a email
~Shaun
"Tom Vile" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
I have a script that will do
lieve that it is on the wiki).
Dovid
--- Miles Scruggs <[EMAIL PROTECTED]> wrote:
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailm
um ok, well do you mind posting it off list to myself, if you haven't
caught it I am interested.
Miles
Tom Vile wrote:
There is a reason why I am posting it off list and not because of money.
On 4/2/06, Miles Scruggs <[EMAIL PROTECTED
eh? what have I done that is rude, I never even made comments about money?
Tom Vile wrote:
um ok, maybe not since you seemed a bit rude.
On 4/2/06, Miles Scruggs <[EMAIL PROTECTED]> wrote:
um ok, well do you mind posting it off list to myself, if you haven't
caught it I am
Great thanks
Steven Job wrote:
OK, enough of this.. No reason to bicker about something like
this.
Here is the URL.
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+cmd+Dial&diff=57
For those of you that do not have a working web browser or cand find
it with Google he
how do you set two types of caller id one for internal calling and one
for external? Basically everyone calling out from asterisk from one
context I want to assign a single callerid. On all other contexts I
want to assign a caller ID specific to each line for all calls going out
to asterisk.
How do you tell if an ext/SIP account.is in use? For instance can you
tell from a Dial() what the status of the line was.
I would like to differentiate between in use and unanswered (for someone
who is on the phone, but doesn't want to take the call)
&
ignored/dnd/unanswered but not in use (f
ccountcode parameter in my sip peer
definitions to the external caller id I want to show, and then I force
the caller id to the ${CDR(accountcode)} variable before placing
external calls.
I don't know if there are any other more efficient methods.
- Waldo
On Apr 6, 2006, at 3:02 AM,
There is plenty of information on the wiki for setting asterisk up for
transferring calls both from the Dail() command, and features.conf.
What really seems to be missing, is simply how do you actually perform
the transfer?
Blind transfers are pretty simple as you only have two obvious steps.
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up from there.
What would be really great is if I could control how many calls by the
context. So if a call was
Benoit Panizzon wrote:
On Sunday 09 April 2006 06:02, Miles Scruggs wrote:
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up from there.
I suppose
C F wrote:
use groups, check the commands/functions group and checkgroup.
I guess I can see how this would be useful, but is there no way to get
it to return BUSY in DIALSTATUS var?
On 4/9/06, Miles Scruggs <[EMAIL PROTECTED]> wrote:
For multiline phones how do you set SIP ch
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
http://72.
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
http://72.1
Steve Totaro wrote:
Miles Scruggs wrote:
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No
application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
http:/
Snip..
Thanks
Miles
If you type "modprobe zaptel" "modprobe ztdummy" at the Linux CLI,
what do you get?
Nothing, they were loaded before, and loaded just fine.
lsmod Module Size Used by
ztdummy 2608 -
rtc106
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
http:/
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
http://7
I'm having issues getting meetme to work:
Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper:
No application 'MeetMe' for extension (internal, , 2)
== Spawn extension (internal, , 2) exited non-zero on
'SIP/mileslap-569b'
the only thing I could find was this:
http://7
I just installed the script, it seems to hang while going out to the
web. Is there someway to have it run in the background while a
background() is playing or something like that?
Thanks
Miles
Jay Milk wrote:
Michelle,
1. Courtesy would suggest that you would have contacted the author of
Ok works like a charm now. So now when I dial ext , I get this:
Created MeetMe conference 1023 for conference '0'
So my question would be, how do I get other people to join this
conference? The voice prompts only tell me that "You are entering
conference number X" where X is 0,1,2
Snip..
you could try dialing from another phone and to dial
either 1023
or 0, my guess is 1023 is what the other people will have to dial.
I would assume that it would work like that, but nope.
from a different phone just creates a new conf, and 1023 is
nev
I made an edit to the wiki:
http://www.voip-info.org/wiki/view/Asterisk+tips+campon
While I need this solution, and I think that some other people can
benefit from various aspects of it, can anyone see if there is a more
elegant solution to achieve the same result? Please feel free to edit
t
Just wondering if anyone has had any luck getting the cisco 7935 working
with asterisk and if so, what is the best way to go about it? on the
wiki there is talk about new software images etc, but I'm thinking those
are for the 7940 & 60 phones. If someone could point me in the right
direction
Just wondering if anyone has had any luck getting the cisco 7935
working with asterisk and if so, what is the best way to go about
it? on the
My testing shows it was a wasted purchase. Using CHAN_SCCP I was able
to get it to work, but not stably (i.e. keys stopped functioning,
phone lo
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