On Sun, Nov 15, 2009 at 06:52:22AM +, aster...@opensourcesolution.in wrote:
i have installed database POSTGRESQL for storing call details. when i
restart database i get the error.
[r...@localhost server]# psql -h
127.0.0.1 -U asterisk Password
psql: could not connect to server:
On Sun, Nov 15, 2009 at 06:27:38AM +, hadi motamedi wrote:
Sorry . I tried to install gcc but I got the following error :
#apt-get update
#apt-get install gcc
You probably meant:
apt-get install build-essential
This installs the basic packages needed for building.
E:Package gcc has
Le samedi 14 novembre 2009 à 20:44 +0100, Eric van der Vlist a écrit :
and playing with udevadm test, I have noticed that the following rule:
+++
KERNEL==[0-9]*, GOTO=xpp_usb_add_end
+++
is always met and
Thank you for your reply . Please find below the required data :
#cat /etc/apt/sources.list
#deb file:///cdrom/sarge main
deb cdrom :[Debian GNU/Linux 3.1 r1a_Sarge_-Official
i386 Binary-1 (20051224)]
/unstable contrib main
deb http://security.debian.org/stable/updates
On 2 Nov 2009, at 12:11, ABBAS SHAKEEL wrote:
Hello,
Let me explain a scenario
There are different Asterisk Servers at different Remote locations.
Recording in different formats for FIVE seconds reveals that
Format : Size
wav : 84 KB
gsm : 8.3 KB
sln : 84 KB
It can be
Thanks Alot all. Specially Tim
It seems to be really good. I will check it in detail
On Sun, Nov 15, 2009 at 3:44 PM, Tim Panton t...@westhawk.co.uk wrote:
On 2 Nov 2009, at 12:11, ABBAS SHAKEEL wrote:
Hello,
Let me explain a scenario
There are different Asterisk Servers at different
hi friends,
i had installed postgres database for call log,call detail.
it has restarted succesfully but when i check tcp connection i dont get any
welcome message by psql.
[r...@localhost ~]# # psql -h 127.0.0.1 -U
asterisk password
[r...@localhost ~]#
th in
aster...@opensourcesolution.in wrote:
*this is my /var/lib/pgsql/data/postgresql.conf*
Tzafrir,
In my client, it didn't show in all upper case (SeaMonkey)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty
On Sun, Nov 15, 2009 at 06:29:21AM -0500, Doug Lytle wrote:
aster...@opensourcesolution.in wrote:
*this is my /var/lib/pgsql/data/postgresql.conf*
Tzafrir,
In my client, it didn't show in all upper case (SeaMonkey)
Interesting. Here's what I see in the HTML part:
pnbsp;strongthis is my
On 15 Nov 2009, at 11:39, aster...@opensourcesolution.in
aster...@opensourcesolution.in wrote:
hi friends,
i had installed postgres database for call log,call detail. it has restarted
succesfully but when i check tcp connection i dont get any welcome message
by psql.
On Sun, Nov 15, 2009 at 10:41:55AM +, hadi motamedi wrote:
Thank you for your reply . Please find below the required data :
#cat /etc/apt/sources.list
#deb file:///cdrom/sarge main
deb cdrom :[Debian GNU/Linux 3.1 r1a_Sarge_-Official
i386 Binary-1 (20051224)]
Tzafrir Cohen schrieb:
On Sun, Nov 15, 2009 at 06:52:22AM +, aster...@opensourcesolution.in
wrote:
THIS IS MY
/VAR/LIB/PGSQL/DATA/POSTGRESQL.CONF
Small hint: Text in ALL CAPS is generally considered as shouting. Please
try to avoid that if you don't really need it.
Doug Lytle
On Sun, Nov 15, 2009 at 1:29 PM, Eric van der Vlist v...@dyomedea.comwrote:
Le samedi 14 novembre 2009 à 20:44 +0100, Eric van der Vlist a écrit :
and playing with udevadm test, I have noticed that the following rule:
+++
KERNEL==[0-9]*,
Gentlemen,
I am trying to find a solution for running a VX-510 over SIP.
I know they have a BTB box that u can use for that purpose but it is, at
least in Sweden,
very expensive.
What I would like to do is something like below.
VX-510 -- SPA2102 -- Asterisk --H.323 trunk-- Avaya CM -- PSTN
Hi
i have a small problems on two Asterisk Server 1.6.4 :
The first sent the call to the second, and in the second, i have a error :
[Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call
rejected, CallToken Support required. If unexpected, resolve by placing
address
Alex Balashov wrote:
It does not appear that you have PostgreSQL set up to listen on a TCP
socket, but only UNIX domain socket. You have this line commented out:
#listen_addresses = 'localhost'
It is required in order to listen on TCP. You should uncomment it:
Unless its not possible with your credit card processor, I would recommend
switching to the ethernet version of the vx-510no hassle and faster
processing.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
_
From:
I know, we can attach something called btb-box (encrypt tcp/ip package)
at the vx-510 and run the transactions over ethernet, I have tested it and
ofc it works but...
Our credit card processor charge us around 15 dollar per month for the
btb-box.
We need one btb-box per office, we have 20
Dear All
In a call flow( dial plan or agi ) , after completion of dial command ,can
we have some information if call is disconnected by called or calling party?
___
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asterisk-users
Phibee Network Operation Center wrote:
Hi
i have a small problems on two Asterisk Server 1.6.4 :
The first sent the call to the second, and in the second, i have a error :
[Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call
rejected, CallToken Support required. If
After a migration to asterisk 1.6, I don't receive sip incoming calls
anymore.
As fas as I understand the SIP debug traces, my server receives the
request and reject it:
++
--- SIP read from UDP:212.27.52.5:5060 ---
I can not help you much, but only confirm that SIP call from one of my provider
in Poland is not working.
Registration goes through OK but call does not go through.
Back to 1.4 version is the solution.
--
Joseph
On 11/15/09 19:05, Eric van der Vlist wrote:
After a migration to asterisk 1.6, I
i am going to set up asterisk for pbx purpose in my office. i am having 2
PSTN lines and will be configuring 10 extentions in my office. plz tell me
which hardware will be needed for this.
thx___
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On Nov 15, 2009, at 2:27 PM, aster...@opensourcesolution.in wrote:
i am going to set up asterisk for pbx purpose in my office. i am having 2
PSTN lines and will be configuring 10 extentions in my office. plz tell me
which hardware will be needed for this.
thx
Have you read this page?
i am going to set up asterisk for pbx purpose in my office. i am having 2 PSTN
lines and will be configuring 10 extentions in my office. plz tell me which
hardware will be needed for this.
Can someone please throw that moron of the list??
apt-get install build-essential
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- hadi motamedi motamed...@gmail.com escreveu:
Sorry . I tried to install gcc but I got the following error :
#apt-get update
#apt-get install gcc
thanks a lot fred for the link.___
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Is there a way to ensure that the source IP address from witch the SIP user
register is not tampred with , is there a feild in the SIP register message
header can be used to achive this security ?
i have an asterisk server in witch SIP users register through an SBC(session
border controller)
Eric van der Vlist wrote:
After a migration to asterisk 1.6, I don't receive sip incoming calls
anymore.
As fas as I understand the SIP debug traces, my server receives the
request and reject it:
++
--- SIP read
Are you referring to the source address of the SIP REGISTER request
itself? If so, you can constrain that, but it would be fairly useless
to spoof it in the general sort of way in which all IP spoofing is
fairly pointless except in a few very particular scenarios, because
the reply will not
Leif,
Le dimanche 15 novembre 2009 à 16:18 -0500, Leif Madsen a écrit :
I'm not sure you've provided enough of the trace here. It finds the peer, but
rejects it with a 401 Unauthorized, which is not uncommon. And I don't see
any
authentication information in the first INVITE. This is why
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a
hotdesk type system where anyone can log on to an extension - however what
I would love to do is relabel the phone with the current owner when this
logon happens. I know that I can change the sip.conf and phones tftp file,
Sometimes they reboot when you try this, but usually not - but you can just
change one setting in the network configuration (eg: change the phones IP
address), and it will go through just the very last part of it's normal boot
process, and re-pull it's TFTP configuration, and update things -
Leif,
Le dimanche 15 novembre 2009 à 22:44 +0100, Eric van der Vlist a écrit :
Leif,
Le dimanche 15 novembre 2009 à 16:18 -0500, Leif Madsen a écrit :
I'm not sure you've provided enough of the trace here. It finds the peer,
but
rejects it with a 401 Unauthorized, which is not
Le dimanche 15 novembre 2009 à 23:45 +0100, Eric van der Vlist a écrit :
Weirdly, they seem to be coming from the context I am using to define
outgoing calls rather than the one for ingoing ones (like in asterisk
1.4), but I guess that's another issue!
Hmmm... I wonder where it can be
On Nov 15, 2009, at 2:27 PM, aster...@opensourcesolution.in wrote:
i am going to set up asterisk for pbx purpose in my office. i am having
2 PSTN lines and will be configuring 10 extentions in my office. plz
tell me which hardware will be needed for this.
On Sun, 15 Nov 2009, Andreas
Throwing him off the list would not achieve anything - he still has our email
addresses, and will still be able to send you email.
Unless of course, you pop his email address on the DENY list of your
gateway...*whistles innocently*
From: asterisk-users-boun...@lists.digium.com
Is the a way for IAX2 adapter to detect when the calling party hangs up the
phone?
I have two IAX2 adapter, one is Digium IAXY when I call the IAXY extension and
hang up the phone the IAXY adapter rings about two or three times after the
calling party hangs up the phone.
The new adapter
On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX
asterisk restarts:
[Nov 15 19:00:36] VERBOSE[17013] logger.c: -- Executing
[...@fax-tx-test:1] ESC[1;36;40mNoOpESC[0;37;40m(ESC[1;35;40mSIP/nhi-rive
rside-sip-ESC[0;37;40m, ESC[1;35;40mContext
sean darcy wrote:
On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX
asterisk restarts:
Before I file a bug, is there anything I'm missing?
Does this happen on earlier versions of the 1.6.0 series prior to this release
candidate? I'm curious if this is a regression, or
Julian Lyndon-Smith wrote:
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a
hotdesk type system where anyone can log on to an extension - however
what I would love to do is relabel the phone with the current owner
when this logon happens. I know that I can change the
Which models of cisco phones (i.e 79x0, 79x1, 79x2, etc). And what do you
mean by VLAN issue.
Thanks,
--Warren Selby
On Sun, Nov 15, 2009 at 7:41 PM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
Julian Lyndon-Smith wrote:
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We
Leif Madsen wrote:
sean darcy wrote:
On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX
asterisk restarts:
Before I file a bug, is there anything I'm missing?
Does this happen on earlier versions of the 1.6.0 series prior to this
release
candidate? I'm curious if
On Sun, 15 Nov 2009, Leif Madsen wrote:
However, changing the label is probably not really the right way to go
about this. For example, I have created an Asterisk system for a call
centre that uses hot desking with Polycom phones, and those phones then
use the built in web browser with
Hi,
I want to record the incoming call in asterisk and instead of writing the
stream to a file..i want to write the stream to a pipe or socket.
In Asterisks code, where i need to do the changes..? If anyone have done this
thing before , plz help me out
Thanks is advance..
Thanks
OK,
Now I am responding to myself, because I have figured it out (finally).
It turns out it's a feature of asterisk (at least the older versions).
This is where I found my answer:
https://issues.asterisk.org/view.php?id=9678
So the solution for me was to simply rearrange my sip.conf so my
Named pipes are presented the same way to calling applications as
files, so you can just write to them as though they were a normal file.
UNIX domain sockets and TCP sockets require socket system calls to
connect to them for the purpose you are trying to achieve, which
Asterisk does not offer
aster...@opensourcesolution.in wrote:
thanks a lot fred for the link.
Just in case you happen to be interested in the more established and
uncontroversial aspects of mailing list usage convention:
This should have been posted as a reply into the existing thread
(Hardware Requirement for
see the DTMF method on both phones.
2009/11/14 Ignacio sanfermi...@gmail.com
Ok, thank you very much. I will try to find any information in
asterisk documentation about RTP.
On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Fri, 2009-11-13 at
Is there any app to pipe a stream to a call either a meetme conference
or even a regular call?
Alex Balashov abalas...@evaristesys.com wrote:
Named pipes are presented the same way to calling applications as
files, so you can just write to them as though they were a normal file.
UNIX
cov...@ccs.covici.com wrote:
Is there any app to pipe a stream to a call either a meetme conference
or even a regular call?
Do you mean piping outside audio of some description into a MeetMe
conference?
If so, I do not know if there is a pre-built app, but this can be
achieved relatively
What is asterisk's behavior when faxdetect=outgoing in zapapata.conf?
Does it turn off echo cancellation?
Does it also change the priority to fax in the outgoing context?
Thanks,
Vieri
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I have done a similar exercise for Grandstream GXP2000 phones, to display
the extension number, user name, DND status.
It uses apache, php and mysql. Uses SIP Notify to update the phone's
display. Phone display status changes within 2 seconds.
I published the initial test version at the
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