Thanks Leif,
That cleared up the versioning.. Is there a list of new features in 1.6.x
versus the 1.4.x version?
On Mon, Aug 3, 2009 at 4:34 PM, Leif Madsen wrote:
> Michael Cunningham wrote:
> > Forgive me if this is a FAQ question but I didnt see anything on the
> > website
> > of forum spell
Hi Guys
I am new working with lumenvox products, and unfortunately I had not been
able to install it properly, I follow the steps in lumenvox site and it
looks like it works I mean:
=
[r...@pbx-millenium examples]# ./example 127.0.0.1
Connect
Hello, all. We attempted an upgraded from 1.6.1.1 to 1.6.1.2 today
including upgrading dahdi-linux from 2.1.0.4 to 2.2.0.2 and dahdi-tools
from 2.1.0.2 to 2.2.0. After rebooting, we receive:
Aug 3 17:20:44 pbx01 kernel: BUG: soft lockup - CPU#2 stuck for 10s!
[swapper:0]
Aug 3 17:20:44 pbx01
Michael Cunningham wrote:
> Forgive me if this is a FAQ question but I didnt see anything on the
> website
> of forum spelling out the difference between 1.4.x and 1.6.x
>
> Obviously 1.6.x is in development. Is it stable enough for production use?
> What are the new features being implemented
Forgive me if this is a FAQ question but I didnt see anything on the website
of forum spelling out the difference between 1.4.x and 1.6.x
Obviously 1.6.x is in development. Is it stable enough for production use?
What are the new features being implemented in 1.6.x?
Will Cepstral work with 1.6.
On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> Hello, all. After reading the README, UPGRADE.txt, and a quick tour
> through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
> one simply compiles and installs over the old installation being careful
> to NOT instal
On Mon, 2009-08-03 at 14:52 -0400, John A. Sullivan III wrote:
> On Mon, 2009-08-03 at 14:27 -0400, John A. Sullivan III wrote:
> > On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote:
> > > On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote:
> > > > On Monday 03 August 2009 12:30:1
On Mon, 2009-08-03 at 14:27 -0400, John A. Sullivan III wrote:
> On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote:
> > On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote:
> > > On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> > > > On Mon, 2009-08-03 at 13:04 -0
On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote:
> On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote:
> > On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> > > On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> > > > Hello, all. After reading the
On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote:
> On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> > On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> > > Hello, all. After reading the README, UPGRADE.txt, and a quick tour
> > > through google, is it safe
On Mon, 3 Aug 2009, Ketema Harris wrote:
> my questions are: What is the correct way(or resource to find a way)
> to get a linux firewall to work with SIP so that the NAT issue is not
> an issue ?
Remove all SIP ALG/connection tracking modules and use old fashioned port
forwarding on the router
On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> > Hello, all. After reading the README, UPGRADE.txt, and a quick tour
> > through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
> > one simply comp
On Mon, 2009-08-03 at 13:29 -0400, Ketema Harris wrote:
> I recently did a set up where I replaced a simple D-link home router
> that was having trouble processing a T1's worth of bandwidth with a
> linux machine running iptables. the kernel was 2.6.29-r5 and I chose
> the SIP connection tra
I recently did a set up where I replaced a simple D-link home router
that was having trouble processing a T1's worth of bandwidth with a
linux machine running iptables. the kernel was 2.6.29-r5 and I chose
the SIP connection tracking modules from the menuconfig.
Router worked fine for norma
On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> Hello, all. After reading the README, UPGRADE.txt, and a quick tour
> through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
> one simply compiles and installs over the old installation being careful
> to NOT instal
Hello, all. After reading the README, UPGRADE.txt, and a quick tour
through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
one simply compiles and installs over the old installation being careful
to NOT install the sample files? Thanks - John
--
John A. Sullivan III
Open Source
If this isn't correct, it is "close enough for government work". By binding
to 192.168.1.10, he makes AMI accessible to anyone who can access that IP
address.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ste
A more specific subject may get better responses. Maybe something like
"How to restrict access to AMI to localhost?"
On Mon, 3 Aug 2009, Jerry Geis wrote:
> Trying to get a setup that only responds to connections on the SAME box.
> like 192.168.1.10 So .10 is my server and I only want responses
On Mon, 3 Aug 2009, Jerry Geis wrote:
> I am talking SIP to the ATA and want to send a dial command with any
> number to it.
> Dial(SIP/myata/somenumber) and have that device come off hook and place
> the call to the somenumber provided.
> Do devices like this exist?
A Linksys SPA3102 will give y
I was playing with the AMI today.
Trying to get a setup that only responds to connections on the SAME box.
like 192.168.1.10
So .10 is my server and I only want responses originating from .10 to
answer the AMI.
I set bindaddr to 192.168.1.10
I left it as 0.0.0.0
I set permit to be 127.0.0.1/25
Alex Hermann wrote:
> On Monday 03 August 2009, Asterisk Team wrote:
>> The release of 1.6.1.2 fixes a remote crash security vulnerability in the
>> RTP stack. The related security advisory AST-2009-004 has been released
>> along with this announcement. Please read that advisory for more
>> infor
Hi
Is there anybody here who has tried to interface Asterisk with PMR446 system
(http://en.wikipedia.org/wiki/PMR446) using the native E&M interface ?
We would like to use Amtelco product H.100 (http://xds.amtelco.com/h100.htm
).
Regards,
Pascal
___
--
On Monday 03 August 2009, Asterisk Team wrote:
> The release of 1.6.1.2 fixes a remote crash security vulnerability in the
> RTP stack. The related security advisory AST-2009-004 has been released
> along with this announcement. Please read that advisory for more
> information.
>
> For a full lis
On Mon, Aug 3, 2009 at 5:08 PM, Jerry Geis wrote:
> I have used the handytone 488 from grandstream in the past
>
> However I need to be able to send a number to a unit like the 488 and
> have it dial out.
> Is there a unit like this available? Basically a 488 unit that can place
> a call out.
>
> >/ I have used the handytone 488 from grandstream in the past
> />/
> />/ However I need to be able to send a number to a unit like the 488 and
> />/ have it dial out.
> />/ Is there a unit like this available? Basically a 488 unit that can place
> />/ a call out.
> />/
> />/ Jerry
> />/
>
On Mon, Aug 3, 2009 at 10:08 AM, Jerry Geis wrote:
> I have used the handytone 488 from grandstream in the past
>
> However I need to be able to send a number to a unit like the 488 and
> have it dial out.
> Is there a unit like this available? Basically a 488 unit that can place
> a call out
I have used the handytone 488 from grandstream in the past
However I need to be able to send a number to a unit like the 488 and
have it dial out.
Is there a unit like this available? Basically a 488 unit that can place
a call out.
Jerry
___
-- B
Pascal Bruno wrote:
> Well I think thats what the problem was, I dont have it named as eth0.
> So if your NIC is not labeled eth0 you cannot use skypeforasterisk???
> Why cant it just scan you nic handles? Can someone point me to where I
> can change the NIC name in the source file or somethi
Steve Totaro a écrit :
> On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel <
> guillaume.yziq...@citycable.ch> wrote:
>
>> Hello.
>>
>> I've set up and configured an Asterisk server to make SIP phone calls to
>> external classic phones.
>>
>> However, it happens that after 15 or 30 seconds, the p
Philipp Kempgen schrieb:
> Elliot Murdock schrieb:
>
>> I am wondering how the Asterisk community has been working on
>> solutions to deal with the asymmetric quality of ADSL. Voip is
>> becoming popular and a bottleneck does exists on the ADSL upload side.
>>
>
> One participant's upload
New never used
bought by mistake ( 5V pci slot ) and stored...
pls make offer,
thanks and regards,
Jean-louis Curty
The TE207P is a bundling of our leading TE205P product and our new
VPMOCT064 Octasic DSP-based echo cancellation module. The TE207P is
Digium's and the industry's first two-port d
New never used
bought by mistake ( 5V pci slot ) and stored...
pls make offer,
thanks and regards,
Jean-louis Curty
The TE207P is a bundling of our leading TE205P product and our new
VPMOCT064 Octasic DSP-based echo cancellation module. The TE207P is
Digium's and the industry's first two-port d
On Mon, 3 Aug 2009, Elliot Murdock wrote:
> Hello Everyone!
>
> Thank you for all the information.
>
> I am wondering how the Asterisk community has been working on
> solutions to deal with the asymmetric quality of ADSL. Voip is
> becoming popular and a bottleneck does exists on the ADSL upload
On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel <
guillaume.yziq...@citycable.ch> wrote:
> Hello.
>
> I've set up and configured an Asterisk server to make SIP phone calls to
> external classic phones.
>
> However, it happens that after 15 or 30 seconds, the phone call drops.
> The SIP session
Hello.
I've set up and configured an Asterisk server to make SIP phone calls to
external classic phones.
However, it happens that after 15 or 30 seconds, the phone call drops.
The SIP session still seems valid, but no sound comes through any more.
How would you go through to troubleshoot thi
On Mon, Aug 3, 2009 at 7:59 AM, Kevin P. Fleming wrote:
> David Backeberg wrote:
>
>> Because of this behavior, and because you want to disable the reinvite
>> for normal audio calls, I think you will have a to use a separate SIP
>> trunk for faxing. I can't think of any other way to do it. So a se
"Kevin P. Fleming" writes:
> No, this is not correct. In spite of its name 'canreinvite' being set to
> 'no' does *NOT* disable all reINVITE operations. It *only* controls
> Asterisk generating reINVITEs for the specific purpose of setting up a
> direct media path. If a reINVITE is needed to swi
Elliot Murdock schrieb:
> I am wondering how the Asterisk community has been working on
> solutions to deal with the asymmetric quality of ADSL. Voip is
> becoming popular and a bottleneck does exists on the ADSL upload side.
One participant's upload is the other participant's download and
vice-
That is up to you. I have never really looked at the pluses.
Personally, it is not for me, especially if you have to do a bit of
tweaking.
Thanks,
Steve
On Sun, Aug 2, 2009 at 9:00 PM, Tarek Sawah wrote:
> do you suggest buying a licensed Software from Digium?
>
>
> -
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Darrick Hartman wrote:
> upgrade-run-image check http://mirror.astlinux.org/firmrware
Note the typo: firmrware
The working command is:
upgrade-run-image check http://mirror.astlinux.org/firmware
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5
I'm not sure there IS an issue, per se. There are lower bitrate codecs
that will work fine for voice communications in both directions. But if
you're trying to force a low-end codec to the upstream, that just means
the downstream on the remote end is going to be stuck with a low-end
codec. And if h
David Backeberg wrote:
> Because of this behavior, and because you want to disable the reinvite
> for normal audio calls, I think you will have a to use a separate SIP
> trunk for faxing. I can't think of any other way to do it. So a second
> parallel SIP trunk for each location sounds like the yo
Hello Everyone!
Thank you for all the information.
I am wondering how the Asterisk community has been working on
solutions to deal with the asymmetric quality of ADSL. Voip is
becoming popular and a bottleneck does exists on the ADSL upload side.
Elliot
On Sun, Aug 2, 2009 at 3:17 PM, Kevin P
Hello Mark,
I managed to make it work - see my previous post
Since you have those phones - does:
voip_control_port: 5060
start_media_port: 1
end_media_port: 10050
works in your case? I tried to put those in SIPDefault but looks like
the phone ignores those and always says:
start media port
Thanks Guys,
I managed to get it working the problem was NAT;
in the sip.conf
[general]
nat=yes
however in the SIP.cnf there was nothing about NAT.
It took me a while to spot it since both asterisk and phone were in
same network and I did not think about NAT.
Solutions:
1) add in sip.conf in [
In wireshark or ethereal:
filter -> sip || rtp
Regards,
2009/8/3 Timothy Weidner
> To make your life a little easier, you can use the following filter:
> sip or sdp or rtp
>
> Just insert that into the filter query field in wireshark and it'll show
> you what you need.
>
> On Sun, Aug 2, 2009 a
Hello D Tucny,
Your solution works indeed well, thanks for it:)
pepesz
Monday, August 3, 2009, 6:20:39 AM, you wrote:
2009/7/31 pepesz76
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my extensions.conf
...
That's interesting. I was always under the impression from what I read
that T.38 was an unreliable, experimental crap-shoot at best and
something that should be avoided for production systems - that the only
reliable solution for FAX was still PSTN lines. Is this no longer true
and all the dire c
Hi,
This is the error message I get. Any idea where I may find some further
debug logs?
> [Aug 3 08:01:23] ERROR[23831] chan_skype.c: Unable to start Skype For
> Asterisk library.
Thanks,
Emrah
Tim Panton wrote:
> I don't know then. My understanding is that the message is caused by
> the wrong
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