Bruce Reeves wrote:
I had similar issues in a deployment where we had set the lease time very
short to aid in getting a new DHCP scope active and remove the older one. I
had not linked the two until you mentioned it, but after we switched
back to
our standard lease time everything worked.
On 1
Is there any way to run a script and or agi that
looks on asterisk and looks for calls that are connected longer X amounth of
time and hang up on them and or look for calls that have not been bridged with a
client within X amount of time and dump the call ?
Thanks.
Dovid
__
Issue ended up being that clients modem was doing
ppoe and it wasnt able to handle more than one phone at a time. As soon as I had
him set the router to do ppoe it worked like a charm.
- Original Message -
From:
Dovid
B
To: asterisk-users@lists.digium.com
Sent: Wed
How about T1? or BRI? I think BRI is just a single channel version of PRI,
right?
Thanks.
King
-原始郵件-
寄件者: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 代理 C F
寄件日期: Sunday, 5 November, 2006 12:51
收件者: Asterisk Users Mailing List - Non-Commercial Discussion
主旨: Re: [asterisk-users] Need
Hi Brian,
I'm sure some other people will give you better answers but quick
answers are;
1/ Depends on volume of message leaving/collection, is it in a single
location? Multiple locations with multiple time zones?
Estimate the number of voicemails left per hour and reply with this.
2/ retrieve
I am totally ignorant about actually using asterisk for any purpose. I
have read some of the docs but not all. I am currently doing a telephone
audit for my company and one of the issues is voice mail. We are spending
quit a bit of money with our telco for voice mail services and I was
wonder
Analog has no way of doing that, unless you listen to the recorded
message and your provider plays the right message for the right cause.
A PRI will give you the cause.
On 11/4/06, King Ho <[EMAIL PROTECTED]> wrote:
Hi,
I am trying to decide on which card to buy for a project where we need to b
In my seemingly endless search for the cause of echo on my SPA3000, I
wired it up in the following configuration:
Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2
And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means
that as soon as I pick up the handset I get linked straight thro
Does the GXP-2000 not have its own dialplan? Use that and disable early dialOn 11/3/06, Anthony Kepler <[EMAIL PROTECTED]
> wrote:I am trying to allow users to place outgoing international calls from a
GPX-2000 with "early dial" enabled, connected to Asterisk 1.2.12.1I have the following extension
When you say you answer the call, I assume you have something like this:exten => 5551212,1,Answerexten => 5551212,1,Dial(SIP/provider/10005551212)Try to not answer the call and see if the behviour changes, it could just be your ITSP configuration
On 11/4/06, hugolivude <[EMAIL PROTECTED]> wrote
Hi,
I am trying to decide on which card to buy for a project where we need to be
able to log the reasons for failed calls.
For example, if the number dialed is not valid anymore, like the client has
changed phone number and decommissioned the old number, we need Asterisk to
return a cause of AST
Hi!
On Tue, 31 Oct 2006, Altus Snyman wrote:
Good day
Im look at
http://www.voip-info.org/wiki-Asterisk+GUI
And I see there are a few GUI for asterisk
What do you guys prefer?
I prefer destar. I like the code form and the extension an customization
possibilities.
What is the best and
Hello all
I would like to know your opinions on free GUI used to manage Asterisk.
Which is better?
My setup is quite small, about 15-20 phones. I've seen the liste on
voip-info.
I have to recommend you destar. A simple python web based interface for
the Asterisk PBX. You can find destar co
If its always 2, then its waiting for CallerID or Fax Detection?On 11/4/06, Jordan Novak <[EMAIL PROTECTED]
> wrote:They are in Kewl start now but I have tried groundstart and
loopstart. Waht could i be missing that would cause this. I start with a
Exten=> s,1,answer. I am using three FXS modules
Its at a remote location, at someone's office. I don't know how there local network is setup. But the phone works fine despite the fact that port changed every minute.
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Zeeshan Zakaria wrote:
I have about 20+ phones on a server, all set for registry expiry 1
min. But only this one, with 2 accounts, keeps re-registerting itself.
All the time this is what I see on asterisk CLI and it is kind of
annoying. What only this phone does this and no other. Its on a rem
programming dept wrote:
What happens is that if we terminate calls to carriers who accept
only the g729 codec we get a 503 service unavailable.
are you sure that your carrier will accept g.729? Sometimes they don't
accept under iax2 and do accept under sip... check your debug for more
informa
Yes, that can be done too. FreePBX on top of Asterisk. I haven't used it this way myself, but this way you keep your original asterisk installation.
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Yea. but is the current chan_unicall.c compatible with Asterisk 1.4?
I don't know how much the channel API changed from 1.2 to 1.4.
The patch for the Makefile I guess I could fix it by hand
BarZ
Moises Silva wrote:
libunicall, spandsp, libmfcr2 are independent from Asterisk version,
the onl
They are in Kewl start now but I have tried groundstart and
loopstart. Waht could i be missing that would cause this. I start with a
Exten=> s,1,answer. I am using three FXS modules on a
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He is not talking about Trixbox but FreePBX and his assumption is correct. Just load Asterisk and then FreePBX later.On 11/4/06, Zeeshan Zakaria <
[EMAIL PROTECTED]> wrote:No, you can't add this to asterisk install. Trixbox formats the drive and installs everything from scratch. But it installs al
I have about 20+ phones on a server, all set for registry expiry 1 min. But only this one, with 2 accounts, keeps re-registerting itself. All the time this is what I see on asterisk CLI and it is kind of annoying. What only this phone does this and no other. Its on a remote location. All phones are
Hi,
We added more g729 licenses today that we bought from digium on Friday.
Since we had previously installed their licenses and had them running we
just ran the register command along with the new key that was given to us.
After we restarted asterisk NONE of the g729's worked even though if we do
Seems likes I am the only person in Asterisk world with this problem, everybody else is fine with audio.
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No, you can't add this to asterisk install. Trixbox formats the drive and installs everything from scratch. But it installs all the necessary software packages which one may need for a perfectly working phone system, and configures them. This saves a lot of time and headache and you can move on fro
Hi!I want to tell asterisk to
simply pass-through any codecs that my phones support. I have to use
codecs that are not popular and implemented by a third-party, asterisk
has nothing to do with them.
I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300
phones to s
Asterisk 1.2.7
RedHat 9.0
I frequently have the need to redirect calls that come in on a DiD
provisioned by my ITSP, back to the ITSP so that they can terminate
the call on the PSTN. For example when an external call comes in, I
often have to send it to a cell phone. I believe that this is
refe
Stephen Bosch wrote:
> Moises Silva wrote:
>> try enabling DTMF debugging on logger.conf for the console, and tell
>> us here waht do you see
>
> This is what comes out on the console, with IP addresses removed:
>
>
>> -- Call accepted by xx.xx.xx.xx (format ulaw)
>> -- Format for call i
Asterisk 1.2.7
RedHat 9.0
I frequently have the need to redirect calls that come in on a DiD
provisioned by my ITSP, back to the ITSP so that they can terminate
the call on the PSTN. For example when an external call comes in, I
often have to send it to a cell phone. I believe that this is
refe
Rob Hillis<[EMAIL PROTECTED]> Wrote on: 11/4/2006 6:46 AM:
> Trixbox isn't a GUI - it's a complete Linux distribution. The GUI to
> Asterisk that comes with Trixbox is FreePBX (http://freepbx.org/)
>
I've been looking for a GUI to use with asterisk. Your response implies
to me that FreePbx can
Hi, Tzafrir,
You're a life saver. After modifying my modules.conf, to load
res_config_mysql.so, app_prepaid.so also loaded. Thanks alot, and to
you to Rodrigo.
Cheers!
On Sat, 2006-11-04 at 23:44 +0200, Tzafrir Cohen wrote:
> On Sat, Nov 04, 2006 at 11:07:52PM +0200, Mosiuoa Tsietsi wrote:
> >
I had similar issues in a deployment where we had set the lease time very short to aid in getting a new DHCP scope active and remove the older one. I had not linked the two until you mentioned it, but after we switched back to our standard lease time everything worked.
On 11/4/06, Doug Lytle <[EMAI
On Sat, Nov 04, 2006 at 11:07:52PM +0200, Mosiuoa Tsietsi wrote:
> Hi again,
>
> I downloaded the bristuff-0.3.0-PRE-1s.tar.gz archive from
> http://www.junghanns.net which has a script you can run to download the
> sources for asterisk (1.2.10), libpri (1.2.3) and zaptel (1.2.6). It
> also has p
Hi Rodrigo,
Thanks for the prompt response. I'm one step ahead of you (sorry for
omitting this info too!).
On my Fedora Core 5 box, I run
$ yum list | grep mysql and get:
libdbi-dbd-mysql.i3860.8.1a-1.2.1
installed
mysql.i386 5.0.22-2.1
installe
Install mysql devel package (depend on your distribution) and will work,
it's not finding the library libmysqlclient
Mosiuoa Tsietsi wrote:
Hi again,
I downloaded the bristuff-0.3.0-PRE-1s.tar.gz archive from
http://www.junghanns.net which has a script you can run to download the
sources for
>> kind of IP phones were you using?
Polycoms
Doug
-- Ben Franklin quote: "Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor Safety."
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Hi again,
I downloaded the bristuff-0.3.0-PRE-1s.tar.gz archive from
http://www.junghanns.net which has a script you can run to download the
sources for asterisk (1.2.10), libpri (1.2.3) and zaptel (1.2.6). It
also has patches for the above as well. Another script helps build the
sources for eac
-Original Message-
>> From: "Scott Keagy" <[EMAIL PROTECTED]>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> Date: Sat, 4 Nov 2006 15:21:08 -0500
>> Subject: RE: [asterisk-users] PRI issues
>> I'm assuming the drop was caused by IP phones renewing leases when the
>
I'm assuming the drop was caused by IP phones renewing leases when the
lease expired (and not handling traffic during the renewal period). What
kind of IP phones were you using?
If my assumptions are off, please clarify the role of DHCP in this
issue.
Thanks,
Scott
-Original Message-
On Sat, Nov 04, 2006 at 02:24:24PM +0200, Mosiuoa Tsietsi wrote:
> Hi all,
>
> I am running Fedora Core 5 on an Intel Server board with 2GB memory and
> kernel 2.6.18-1.2798.fc6 . I am running asterisk-1.2.10 from
> bristuff-0.3.0-PRE-1s. I have a mysql-version of app_prepaid which
> builds prope
Moises Silva wrote:
> try enabling DTMF debugging on logger.conf for the console, and tell
> us here waht do you see
This is what comes out on the console, with IP addresses removed:
> -- Call accepted by xx.xx.xx.xx (format ulaw)
> -- Format for call is ulaw
> -- IAX2/[provider_chan
Hi,
After upgrading from:
Zaptel 1.2.9.1
Asterisk 1.2.12.1 with bristuff-0.3.0-PRE-1s
to
Zaptel 1.2.10
Asterisk 1.2.13 with brustuff-0.3.0-PRE-1v
I get the following error when connecting my Xlite Softphone:
--- cut ---
Nov 4 17:33:45 WARNING[4430]: chan_sip.c:1090 __sip_xmit: sip_xmit of
0x886d
I guess you have done these steps already
modprobe zaptel
modprobe wcfxo
ztcfg -vv
If the above steps didn't help move the x100p card to different PCI slot, if everything is occupied, swap with some other pci card
Add the defaultzone to the zaptel.conf.
let me know if this helps.
O
I think you can use asterisk for your sip implementation.
As you already know, SIP is peer to peer and you don't need asterisk when u want to communicate just between two users. If you are thinking about having more than 2 users (say 25) in the setup, Asterisk will help you there. In this case, A
Mike wrote:
PS: If there is a better FTP server suggestion Ill take it, but one of
my "must-haves" is easy of use and virtual users functionality (with
different chroot folders).
I don't know whether it supports the specific functionality you require,
but we have always uses vsftpd with no p
Doug Lytle wrote:
Hey everybody,
I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T.
I've received several complaints about dropped calls. Reviewing the
archives on PRI and dropped calls shows that I should set the
resetinterval=never in the zapata.conf and restart. This
J
Can anyone confirm if asterlink is down? We are using SIP
connectivity to them and they seem to have been down since at least
3AM EST. It is not 1030AM EST, and they are still down. I can get no
responce from their support desk. Can anyone offer any insight?
__
Unless you need to provide provisioning information outside your
network, I suggest using TFTP instead of FTP.
Matt
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Just for the record.. asterisk was held up as it was trying to resolve
a DNS name for a sip peer that that didn't have a correct DNS entry.
On 11/4/06, Matt <[EMAIL PROTECTED]> wrote:
Hi,
I am trying to upgrade my system (running 2.4 kernel) from 1.0.9 to
1.2.6, everything upgraded fine, however
Asterisk 1.2.7
RedHat 9.0
I frequently have the need to redirect calls that come in on a DiD
provisioned by my ITSP, back to the ITSP so that they can terminate
the call on the PSTN. For example when an external call comes in, I
often have to send it to a cell phone. I believe that this is
refe
Response inline.
Steve Totaro wrote:
I receive calls over a T1 with callerid and then *ani*dnis*. I am able
to strip out the ani and the dnis in the dialplan but when I try to set
the caller ID to be the ani, it looks ok but then if I do a NoOp
callerid on the next line, I get unknown.
Here
Hi all,
I am running Fedora Core 5 on an Intel Server board with 2GB memory and
kernel 2.6.18-1.2798.fc6 . I am running asterisk-1.2.10 from
bristuff-0.3.0-PRE-1s. I have a mysql-version of app_prepaid which
builds properly but when I run asterisk -gc I get the following:
[EMAIL PROTECTED] /]# a
Trixbox isn't a GUI - it's a complete Linux distribution. The GUI to
Asterisk that comes with Trixbox is FreePBX (http://freepbx.org/)
Zeeshan Zakaria wrote:
Trixbox
www.trixbox.org
signature.asc
Description: OpenPGP digital signature
_
Hi,
I'm cutting my teeth setting up a home asterisk server, but having some
trouble getting Zaptel to recognize a X100P clone PCI (Motorola SM56
which should work according to various wiki's).
What struck me was that the "/proc/interrupts" do not include IRQ 11 as
the "lpcpi -v" is reporting
Trixbox
www.trixbox.org
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I second that request.
On 11/4/06, Kevin Bockman <[EMAIL PROTECTED]> wrote:
Hi,Would anyone be kind enough to send me the 2.0.2 SIP firmware? I askedVoipSupply for it on Wednesday, nagged them again on Thursday and theydid not even send the request yet. I was supposed to have it 'Friday
morn
Hi Kevin,
you have to create a gateway in the Smart Node:
gateway sip sip
bind interface eth1 router
service default
domain gwsnettech.local
realm gwsnettech.local
authentication isdngw2 password huffvtzddzdjkhuztztufuz== encrypted
default
default-server hallinux2.gwsnettech.
I would like to ask, if someone observe also problem with peer qualify
problems,
my asterisk log is full with UNREACHABLE/REACHABLE messages, even when
two asterisks are in LAN environment,
please take a look into this debug, I can't find any problem with packet
loss, all qualify requests are re
need help on this..
I have configured all my internal extensions as SIP phones, i have one
ATA-186 and one softphone. When I try to transfer calls between
internalt, I use *1 as it configured on feature.conf.
But my external line configured a IAX.
When there is an incoming call from outside, th
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Crazy Boy wrote:
> Hi Friends,
>
> I have an account with sipdiscount.com. I configured my Asterisk server.
> When I try to make a call, its telling that "All circuits are busy". I
> tried in many ways. Can anybody send me correct working configuratio
Hi,
I am trying to upgrade my system (running 2.4 kernel) from 1.0.9 to
1.2.6, everything upgraded fine, however asterisk is not seeing any
zap/sip/iax2 channels.
I compiled in this order: libpri/zaptel/asterisk. Zaptel comes up
fine... ztcfg -vv shows all of my channels, however asterisk lacks
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