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
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
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 pepes...@o2.pl
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my
In wireshark or ethereal:
filter - sip || rtp
Regards,
2009/8/3 Timothy Weidner tweidner6...@gmail.com
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
Thanks Guys,
I managed to get it working the problem was NAT;
in the sip.conf
[general]
nat=yes
however in the SIPMAC.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
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
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
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 your
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
-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
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 tareksa...@hotmail.com wrote:
do you suggest buying a licensed Software from Digium?
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
Kevin P. Fleming kpflem...@digium.com 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
On Mon, Aug 3, 2009 at 7:59 AM, Kevin P. Flemingkpflem...@digium.com 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
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
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 still
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 side.
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
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
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 is the
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 phone call drops.
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
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 ge...@pagestation.com 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
/ 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
//
//
/I am not sure
On Mon, Aug 3, 2009 at 5:08 PM, Jerry Geis ge...@pagestation.com 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
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 list of
Hi
Is there anybody here who has tried to interface Asterisk with PMR446 system
(http://en.wikipedia.org/wiki/PMR446) using the native EM interface ?
We would like to use Amtelco product H.100 (http://xds.amtelco.com/h100.htm
).
Regards,
Pascal
___
--
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
information.
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
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 you 1
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
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
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
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 install
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
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
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 compiles
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 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 to
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 README,
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 -0400, John
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 12:30:12 pm John
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 install
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
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 in
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
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
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 leif.mad...@asteriskdocs.orgwrote:
Michael Cunningham wrote:
Forgive me if this is a FAQ question but I didnt see anything on the
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