On Sunday 15 November 2009 00:01:13 Jarrod Lash wrote:
> apt-get update
> then
> apt-get install gcc g++
There's also a package on Debian called 'build-essential' which gets all
of the usual libraries and headers such that compiling works right the
first time.
--
Tilghman Lesher
Digium, Inc. | S
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:
listen_addresses = '127.0.0.1'
In any case,
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:
Connection refused
Is the server running on host "127.0.0.1" and
accepting
TCP/IP connecti
you are running a old version of debian?
what repository are you using (cat /etc/apt/sources.list)?
On Sun, Nov 15, 2009 at 1:27 AM, hadi motamedi wrote:
> Sorry . I tried to install gcc but I got the following error :
> #apt-get update
> #apt-get install gcc
> "E:Package gcc has no installatio
Try:
apt-cache search gcc | grep '^gcc'
and pick the more precise package to install.
hadi motamedi wrote:
> Sorry . I tried to install gcc but I got the following error :
> #apt-get update
> #apt-get install gcc
> "E:Package gcc has no installation candidate"
> Can you please do me favor and l
Sorry . I tried to install gcc but I got the following error :
#apt-get update
#apt-get install gcc
"E:Package gcc has no installation candidate"
Can you please do me favor and let me know why ?
Thank you in advance
On Sun, Nov 15, 2009 at 6:01 AM, Jarrod Lash wrote:
> apt-get update
> then
>
apt-get update
then
apt-get install gcc g++
--
Jarrod Lash,
Federated Communications
www.fed-com.com
Office: +1-412-357-2127
Mobile: +1-412-999-0049
Fax: +1-412-545-8368
On Sun, Nov 15, 2009 at 12:31 AM, hadi motamedi wrote:
> Dear All
> Please be informed that I need to install Asterisk 1.4.
Also install a recent version! 1.4.26.3 would be the latest in the 1.4
release series. Using something as old as 1.4.13 is not recommended.
Alex Balashov wrote:
> You need to install 'gcc' and 'g++' and associated libraries and headers.
>
> hadi motamedi wrote:
>
>> Dear All
>> Please be info
You need to install 'gcc' and 'g++' and associated libraries and headers.
hadi motamedi wrote:
> Dear All
> Please be informed that I need to install Asterisk 1.4.13 on my Debian
> 3.1 server . But I got the following message when trying for
> "#./configure" :
> "error: no acceptable C compile
Dear All
Please be informed that I need to install Asterisk 1.4.13 on my Debian 3.1
server . But I got the following message when trying for "#./configure" :
"error: no acceptable C compiler found in $PATH"
Can you please do me favor and let me know what is the problem ?
Let me thank you in advanc
Phibee Network Operation Center wrote:
> Hi
>
> I have a problems with a new Asterisk Server,
>
> when i want call, i have:
>
> [Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
> handle_request_invite: Call from 'PHISIP01' to extension
> '00420225352184' rejected because extension not fo
Peter Evans wrote:
> On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote:
>
>> hii guys:
>> i get the message from the asterisk:
>>Started music on hold, class 'default', on
>> Local/s...@skype-web-callback-dial-263to263-1775,1
>> [2008-11-11 14:32:41] WARNING[1781]: fo
On 11/13/09 18:31, Hans Witvliet wrote:
>On Thu, 2009-11-12 at 20:18 -0700, Joseph wrote:
>> Digium has discontinued their ATA iaxy adapter; don't blame them, too
>> expensive so they can not compete.
>
>Compete, With which iax-ata ???
How about AG-188N
Though, I just notice this unit has a sec
Landy Landy wrote:
> I believe language barriers can cause many problems when trying to
> communicate. I might say something in another language trying to
> translate a phrase or something, that might not have the same
> meaning I´m trying to get accross. I´m billingual myself, english
> is my
On 09/21/09 17:54, Vincent wrote:
>Hello
>
>According to this article, this nice little unit can only use the PSTN
>port for outgoing calls (ie. as a backup in case the connection to the
>VoIP provider stops working), but not incoming calls:
>
>http://tinyurl.com/mwjmo8
>
>Can someone confirm that
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
wrote:
> On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote:
>> I have just established a call between 2 sip phones and I have noticed
>> that all R
> > Pre-judging people doesn't work on mailing lists given
> the
> > inherent language barriers, etc.
I believe language barriers can cause many problems when trying to communicate.
I might say something in another language trying to translate a phrase or
something, that might not have the same
Pawan,
I am getting the sense that you do not understand how to use a mailing
list, so I will take the time to uh, "guide" you before answering your
question as best as I can. u r welcome
The purpose of making a posting to a mailing list is not to find a
single person that replies to you on the
Hi Tzafrir,
Le samedi 14 novembre 2009 à 21:30 +0200, Tzafrir Cohen a écrit :
> On Sat, Nov 14, 2009 at 12:55:46AM +0100, Eric van der Vlist wrote:
> > Hi,
> >
> > I have upgraded an Asterisk installation with a Xorcom BRI Astribank
> > that was working under Ubuntu 8.04 to Ubuntu 9.10 and the de
Bandino Jurumai wrote:
> Can anyone tell me how to specify subroutine call with arguments in the
> Asterisk 1.6 Queue application?
> Documentation does not mention what is the syntax for specifying the
> subroutine with arguments.
If that functionality exists... try using ^ for the separator:
Q
Gavin Spurgeon wrote:
> Thanks to some answers on this list and another I now have a MultiTenant
> system/setup working the way that I want it to, So now my next job is
> to find a SIP SoftPhone that I can brand to my own company images and
> so on.
>
> Again an OSS would be preferred, Even though
On Fri, Nov 13, 2009 at 01:36:00PM -0500, Humanx2000 wrote:
> Just picked up Asterisk the Future of Telephony, every other listed
> program is there (Book does not tell you about the changeover to
> dahdi_). But there is no dahdi_zttools. I have dahdi-tools
> installed, tried to install via yum and
On Sat, Nov 14, 2009 at 12:55:46AM +0100, Eric van der Vlist wrote:
> Hi,
>
> I have upgraded an Asterisk installation with a Xorcom BRI Astribank
> that was working under Ubuntu 8.04 to Ubuntu 9.10 and the device is no
> longer initialized.
>
> When I reload the udev rules, I see that the rules
Don't laugh too hard.
--
Sent from mobile device
On Nov 14, 2009, at 2:04 PM, ABBAS SHAKEEL
wrote:
I cant stop laughing lolz
Any how we must not reply in private but ask to post on list only.
Lets make him able to achieve his objective through the list.
Cheers
On Sat, Nov 14, 2009 at
I have iptables FORWARD to ACCEPT by default:
iptables -P FORWARD ACCEPT
and still have the same problems.
Now, the dsl modem is also opened. not blocking any ports as well.
--- On Sat, 11/14/09, Michelle Dupuis wrote:
> From: Michelle Dupuis
> Subject: Re: [asterisk-users] Can't connect
Thanks. In that case, do me a favour in return and start using the
mailing list as it is intended, instead of mailing people privately.
--
Sent from mobile device
On Nov 14, 2009, at 1:58 PM, wrote:
> hi alex done with music on hold. n thanks a lot for the reply. u r
> the only person who
Ok I am replying to myself, because I still don't have this figured
out,, but I think I have more info.
On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote:
>
> Hello again Asterisk people.
>
> I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
> deployed for several years now,
I cant stop laughing lolz
Any how we must not reply in private but ask to post on list only. Lets
make him able to achieve his objective through the list.
Cheers
On Sat, Nov 14, 2009 at 11:56 PM, Alex Balashov
wrote:
> I don't get it. I just replied helpfully to Mr. "opensourcesolutions"
> on
I don't get it. I just replied helpfully to Mr. "opensourcesolutions"
on the mailing list and for this he expresses his gratitude with two
more obnoxious private e-mails to me, along these general lines:
hi these r the steps to rn music on if i doing some mistake than plz
guide me.
1-
Sorry for causing this war. It's just, if everyone sent private
messages:-
a) there would be no point of the mailing list
b) our mailboxes would fill up in minutes, leaving no space of our
business emails.
Dan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:aste
Hi,
Can anyone tell me how to specify subroutine call with arguments in the
Asterisk 1.6 Queue application?
Documentation does not mention what is the syntax for specifying the
subroutine with arguments.
Thanks
___
-- Bandwidth and Colocation P
I'll start with a guess - your asterisk box or firewall is blocking SIP
ports. Diagnose that first (stop iptables/check iptables if unsafe) and try
again...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Land
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List (Again)
Thanks to some answers on this list and another I now have a MultiTenant
system/setup working the way that I want it to, So now my next job is
to find a SIP SoftPhone that I can brand to my own company images and
so on.
Again an OSS
aster...@opensourcesolution.in wrote:
> i had done r/d of voice mail in which i got succes, now when i call
> exten 2000 and it on hold there is no music on hold. plz guide me what
> mistakes i am doing.
Have you made the necessary adjustments to
/etc/asterisk/musiconhold.conf to define music
hi friends,
as i am a beginner in voip, i had made a very simple dial
plan i had made two extentions n both are able to ring each other through
soft phone (X-Lite)
below is my dialplan
#
Is anybody using AG-188N?
It is a nice little unit, it would be a perfect X-mas gift for your family
member abroad. It supports IAX2/SIP with STUN server.
Both IAX2 and SIP can be registered at the same time to two different
providers. The limiting part is PSTN, it is just a lifeline not a real
On Sat, 14 Nov 2009, Phibee Network Operation Center wrote:
> when i want call, i have:
>
>[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
> handle_request_invite: Call from 'PHISIP01' to extension
> '00420225352184' rejected because extension not found.
(You don't say what version of A
On Sat, 14 Nov 2009, hadi motamedi wrote:
> Can you please do me favor and let me know how can I stop my Asterisk ? Can
> you please confirm if the following procedure is correct to stop it ?
> #/etc/init.d/asterisk stop
Yup.
> #cd /etc/init.d
> #chmod asterisk
These commands imply you do
Update on my problem.
After a few days of speaking with various folks at ATT, the issue has
been "resolved". When we first ordered our PRI lines, they were
supposed to be without screening tables (per ATT suggestion on how we
can set CID numbers to numbers not associated with the PRI). However,
On Sat, Nov 14, 2009 at 10:20 AM, Marcus Vinicius
wrote:
> I'm trying to send faxes using Asterisk 1.4 and T38 with sip but Asterisk
> rejects the t38.
>
> Anybody know if is possible to transmit t38 fax with Asterisk 1.4?
It might be possible to T.38 fax with 1.4, but I do not recommend it.
T.
Hi,
I'm trying to send faxes using Asterisk 1.4 and T38 with sip but Asterisk
rejects the t38.
Anybody know if is possible to transmit t38 fax with Asterisk 1.4?
following settings:
--- sip.conf ---
[general]
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
context=from-outs
According to my provider they´re not receiving any request from us but, now
everytime I try to place a call through them I´m getting:
*CLI> sip show peers
Name/username HostDyn Nat ACL Port Status
100(Unspecified)D 5060 Unmonit
hadi motamedi wrote:
> Dear All
> Can you please do me favor and let me have the link to download the
> Asterisk 1.4.13 for my Debian server ? Please let me know how to install
> it .
> Thank you in advance
Well, 1.4.13 is quite old now, but you can find the older released versions of
Asterisk
Hi All;
I am trying to have the possibility to pass traffic from SIP to H323, and I am
using the asterisk version 1.4.26.2 with h323 (so I have h323.so channel), the
h323 listens at port 1722 TCP and on the same machine I have gnugk running and
listens on 1721 TCP.
When placing a call from the
Hmmm. Let me rephrase your question:
"Dear List: How do I make server b and c do what I want when I have no control
over b or c?"
Enough said.
-K
- Original Message -
From: B.Masoud @ SH
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Thursday, Novemb
if you're using a redhat-derived distro you can always use the easy-to-remember
"service asterisk stop"
-K
- Original Message -
From: ast guy
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Saturday, November 14, 2009 4:12 AM
Subject: Re: [asterisk-users] I
Hmmm. Let me rephrase your question:
"Dear List: How do I make server b and c do what I want when I have no control
over b or c?"
Enough said.
-K
- Original Message -
From: B.Masoud @ SH
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Thursday, Novemb
This one is manual way of doing it.
You can get more details at page
http://www.voip-info.org/wiki/view/Asterisk+Starting+and+Stopping
It has provided the init.d scripts where you can automate the process.
/ag
On Sat, Nov 14, 2009 at 7:44 AM, Yawar Hadi wrote:
> cli> stop now
> or
> cli > sto
I am searching for a GUI platform that can support multiple sites with
Asterisk servers with a single GUI to manage them all.
I have something currently working but it is not as pretty or polished as I
would like, although it does work quite well.
Are there any opensource options out there?
Than
Hi
I have a problems with a new Asterisk Server,
when i want call, i have:
[Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160
handle_request_invite: Call from 'PHISIP01' to extension
'00420225352184' rejected because extension not found.
but into my extensions.conf:
exten => _
Dear All
Can you please do me favor and let me have the link to download the Asterisk
1.4.13 for my Debian server ? Please let me know how to install it .
Thank you in advance
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
aste
51 matches
Mail list logo
