5 sep 2009 kl. 20.02 skrev Jens Wolf:
Hi,
i have try to send IM from Client A (Ekiga) to Client B (Ekiga).
Realtime text is sent in the RTP stream. What you're sending is a SIP
message. Those are two different things.
Asterisk does not support SIP messages outside of a call - so test
2009/9/5 Steve Edwards asterisk@sedwards.com
snip
You can, but I don't. I do it like this:
snip
Very, very interesting practises.
Originally, I opened this thread when I wondered whether or not, I should
centralize logs (and from that original question, I came to study some tiny
Hello!
I have a very simple bash script:
#!/bin/bash
asterisk -rx sip show peers /var/log/devices
When I run it in bash shell, everything works fine, but if I
background it (by adding or using bg), nothing appears in the
/var/log/devices file.
Any reason for this behavior or help would be
From my research:
Our Receiver
ssrc our ssrc
rxcount no. received packets/Received packets
lplost packets/Lost packets
rxjitter our calculated jitter(rx)/Jitter
Our Sender
themssrc their ssrc
txcount
On Sun, 6 Sep 2009, Elliot Murdock wrote:
Hello!
I have a very simple bash script:
#!/bin/bash
asterisk -rx sip show peers /var/log/devices
When I run it in bash shell, everything works fine, but if I
background it (by adding or using bg), nothing appears in the
/var/log/devices file.
Hello Gordon!
Thanks...it works now.
How did you notice that it needed input from the keyboard?
Do you think I should make the developers aware of this issue?
Thanks,
Elliot
On Sun, Sep 6, 2009 at 2:12 PM, Gordon
Hendersongordon+aster...@drogon.net wrote:
On Sun, 6 Sep 2009, Elliot Murdock
On Sun, 6 Sep 2009, Elliot Murdock wrote:
Hello Gordon!
Thanks...it works now.
How did you notice that it needed input from the keyboard?
Experience. (Of being a unix/linux user/sysadmin for 25+ years)
I ran:
dsx# rasterisk -x 'sip show peers' /tmp/foo
[1] 8172
dsx#
Then hit
Hi All,
does Digium provide a service support for a compatibility question about their
PRI hardware ?
Thanks and have a nice day.
___
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AstriCon 2009 - October 13 - 15 Phoenix,
On Sun, Sep 6, 2009 at 10:33 AM, mancyb...@gmail.com mancyb...@gmail.comwrote:
Hi All,
does Digium provide a service support for a compatibility question about
their PRI hardware ?
Thanks and have a nice day.
They do support their cards. Why not call and find out?
--
Senior Systems and
Is there a version of this patch for 1.6.2 - or did the recent 1.6.2
rc1 drop include it ?
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
smime.p7s
Description: S/MIME cryptographic signature
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2009/9/5 Steve Edwards asterisk@sedwards.com
snip
You can, but I don't. I do it like this:
snip
On Sun, 6 Sep 2009, Olivier wrote:
Very, very interesting practises.
Originally, I opened this thread when I wondered whether or not, I
should centralize logs (and from that original
On Saturday 05 September 2009 23:10:19 Luke McKee wrote:
I bought paid for and used to use ChanSkype for skype connectivity.
Now I tried to download the software again after the my old server got
retired and someone else is squatting on their website. Can someone
give me the latest ChanSkype
On Sunday 06 September 2009 12:31:05 Tim Panton wrote:
Is there a version of this patch for 1.6.2 - or did the recent 1.6.2
rc1 drop include it ?
There is. If you look in the security subdirectory on the downloads server,
you'll find a patch for 1.6.2. No, rc1 of 1.6.2 was released too early
I just upgraded to 1.6.2.rc-1 after running betas 2 and 3 with no
problems and while everything seems fine i get these message at
startup and than all is well. Should I be worried or do i need to let
the team know about this?
Also, is not finding /dev/dahdi/transcode a problem I should be
Steve Edwards wrote:
2009/9/5 Steve Edwards asterisk@sedwards.com
snip
You can, but I don't. I do it like this:
snip
On Sun, 6 Sep 2009, Olivier wrote:
snip /
To avoid that, have you tried tools like phpLogCon ?
Nope. Never heard of phpLogCon before.
Since it hasn't come up
On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote:
- Barry Miller asterisk-us...@notanet.net wrote:
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840
work
fine.
With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a
line
I'm attempting to link calls together in my CDR and would like to try to do it
via the userfield. Is there any way to copy the userfield between calls when
doing an attended transfer? I can't seem to find anything about it searching
Google.
-Jon
___
that's probably for ADSI phones ... chan_local confuses the VoiceMailMain app
and you hear it ... Why do you need to call it via chan_local ? Can't
you do Macro or just
call VoiceMailMain directly ?
Martin
On Fri, Sep 4, 2009 at 3:28 AM, Santiago
Gimenosantiago.gim...@gmail.com wrote:
Hello,
The system specs mention PCIe expansion slots, so your only
option is the TE420B.
--
exvito
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
does Digium provide a service support for a compatibility question
about
their PRI hardware ?
Before you open a call with them, you will have to register your Digium
card by entering the serial number. The serial number is printed on a
sticker which is attached to the card. There is no way
Hello team;
While am aware and active user of astersk monitor function for recording, i
would like to know if i can use asterisk as a pure recording server(like nice
or witness) for some other PABX's extensions (both inbound, outbound and
internal).
Setup
PSTN---Legacy PABX(with analogy n
On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote:
Hello team;
While am aware and active user of astersk monitor function for recording, i
would like to know if i can use asterisk as a pure recording server(like
nice or witness) for some other PABX's extensions (both
I have also seen:
PSTN asterisk legacy
Which also gives you a migration path
PaulH
Research wrote:
Hello team;
While am aware and active user of astersk monitor function for
recording, i would like to know if i can use asterisk as a pure
recording server(like nice or
Paul,
He already said, While am aware and active user of astersk monitor function
for recording so I don't think migration path is an issue.
A dedicated recording server is recommended if you are going to be recording
a good deal of calls.
You certainly would not want to run out of hard drive
On Sun, Sep 6, 2009 at 11:46 PM, Steve
Totarostot...@asteriskhelpdesk.com wrote:
A dedicated recording server is recommended if you are going to be recording
a good deal of calls.
You certainly would not want to run out of hard drive space on your Asterisk
server and bring it down.
Bring it
On 5/09/09 6:51 PM, Olle E. Johansson wrote:
5 sep 2009 kl. 01.05 skrev Matt Riddell:
On 4/09/09 6:22 PM, Olle E. Johansson wrote:
4 sep 2009 kl. 00.44 skrev Matt Riddell:
On 4/09/09 10:41 AM, Doug Lytle wrote:
Todd Routhier wrote:
Trying to do something like this in the sip.conf under
Did you push it past 300 on two year old hardware and software? Core 2 Duo
Dell Dimension desktop as proof of concept?
Port mirroring is basic on almost any newer switch. Login, enable port
monitoring, write mem, done.
With a GUI, it takes all of thirty seconds. I don't see how this is heavy
In layman's terms, if you read the OP's post, they want a *recording server*.
With no mention of any other functionality.
Asterisk *COULD* be used as a recording server but that is really not the
proper tool, on the other hand OrecX *IS* a recording server, therefore the
proper tool.
I *COULD*
Ira wrote:
I just upgraded to 1.6.2.rc-1 after running betas 2 and 3 with no
problems and while everything seems fine i get these message at
startup and than all is well. Should I be worried or do i need to let
the team know about this?
Also, is not finding /dev/dahdi/transcode a problem
On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote:
On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote:
Hello team;
While am aware and active user of astersk monitor function for recording, i
would like to know if i can use asterisk as a pure recording
On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote:
On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com
wrote:
Hello team;
While am aware and active user of astersk monitor function
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