Mauro Zanin wrote:
Hi everybody,
I installed a 3.0Gb 512MB TrixBox with a Celeron inside. The PC seems smart
enough an rapidly the CentOs and Asterisk were loaded on it. I has only 2
extensions(SIP telephones, one GPX2000 and one Grandstream 486) and one ISDN
adapter with Bristuff. Web server
Jadrien Wauthier wrote:
Does anyone know how to adjust the volume of the ringing
application? I
have done a lot of internet searching and have not found much.
You cannot do this in Asterisk.
Some SIP phones might allow you to do so by setting an option on the
phone, but you would have to
[EMAIL PROTECTED] wrote:
I experienced the same problem. The only way I could get both
ISDN and analog working was unloading kernel modules for zaptel
and mISDN after boot and then load them in the order:
zaptel first and then mISDN. Still need to find out how to configure
load order in
Douglas Garstang wrote:
Can anyone recommend any wholesale SIP termination providers that will
automatically charge a credit card? Most seem to want upfront payment
and a credit balance but that sucks when you have to watch it like a
hawk to make sure the balance never hits zero. I’m
Bernardo Vieira wrote:
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Gordon Henderson wrote:
If you only have one * box behind the NAT gateway then I don't really
see a big issue with it to be honest. Port-forward on the
firewall/router device (5060 and 1 through 2) to the *
Bernardo Vieira wrote:
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Bernardo,
Just a thought: Try using a different SIP port for one of the
extensions, if possible.
Bob...
Bob,
Tanks for the tip. I had actually done that before, as a matte of fact
that's the solution I have in
Hugh L. Johnson wrote:
I have a Dish 301 receiver that will not display CallerID when connected
to FXS module on TDM400. Uniden phone connected to the same FXS module
does display CallerID.
When Dish 301 receiver is connected to IAXy CallerID is displayed
properly.
Any suggestions on getting
Ronald Wiplinger wrote:
How can I set-up a MRTG with 4 graphs, whereby:
1 data in
2 data out
3 ONLY voice(/video) data in
4 ONLY voice(/video) data out
bye
Ronald Wiplinger
___
--Bandwidth and Colocation provided by Easynews.com --
Steve,
VoiceEclipse has a US unlimited plan for $20/month. Two inbound numbers
that can be in different area codes. I have not figured out how to
recognize which number the inbound call came in on, but, right now, that
is not that important to me. Others have had other problems. Research
Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in:
http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22
By default, if my asterisk went down after the SPA3000 was already
registered, the in-bound PSTN call was lost. I probably did not wait
Elpidio,
Is it truly not listening or is maybe a firewall blocking port 5060.
What does netstat -an | grep 5060 tell you? I get this:
netstat -an | grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
iptables -L will list any firewall restrictions.
Bob...
On Fri, 2006-09-01
REJECT all -- anywhere anywhere
reject-with icmp-host-prohibited
I assume this indicates port 5060 is restricted?
Elpidio
*/Bob Chiodini [EMAIL PROTECTED]/* wrote:
Elpidio,
Is it truly not listening or is maybe a firewall blocking port 5060.
What does
.
Thanks
*/Bob Chiodini [EMAIL PROTECTED]/* wrote:
I think all anywhere should allow 5060. Try running service iptables
stop (as root) to shutdown the firewall. See if 5060 then answers.
I'm not running a firewall on my asterisk box so I'm not sure what
the
rule would need
Nick,
I've used a SPA3000. There seems to be a later model from Linksys,
hopefully it works better. I had some severe echo problems due to my
distance from the CO. The SPA3000 never could seem to compensate. The
older software worked better, but it never passed muster with the wife.
Went to
Nick,
I know some adults that can have an entire conversation in the same
amount of time.
Does pressing the # key speed up dialing? If so look for a timer in the
PAP config or tell the kids to press #. IIRC the spa3k had something
similar, but never did much in-house dialing.
$86 is a
Nick,
Anything helpful in the asterisk or system logs.
Try bumping up the debug and verbose levels see what shows up on the
console.
Weird that it would work inbound and not outbound.
Bob...
On Thu, 2006-09-07 at 04:48 -0700, Nick Ellson wrote:
Hey all,
A previous annoyance with not
Rushowr wrote:
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Hey all,
I'm looking into setting up a system or two with either IMAP or ODBC
storage of Voicemail messages and wanted to hear about your experiences,
gather tips or warnings, etc, before I go diving too deep into it. Are
either of
Does the Linksys know it should be using port 5070? It would seem to me
that port forwarding would be required as the phones are behind a NAT'd
firewall. How would asterisk know how to get there since it's not on
the same subnet (outside the firewall).
If the asterisk box has physical access to
On Mon, 2006-09-11 at 20:25 -0700, Christopher Corn wrote:
im having issues when routing calls from the outside with my new VSP.
this is what asterisk tells me when i try to make an incoming call, i
get the no service response when i call.
-- Executing GotoIf(SIP/christopher_corn-eddb,
On Wed, 2006-10-04 at 14:58 -0700, Naija Man wrote:
Hello all,
Asterisk 1.2.8
zaptel 1.2.6
Hardware: digium TDM2422P
I have a fully configured asterisk system with POTS line for PSTN
access. I am not receiving the callerid for incoming calls from the
PSTN. I get the following error
On Wed, 2006-10-11 at 11:19 -0400, Jerry Geis wrote:
I lost my internet connection today for a short time.
During that time 1.2.12.1 stopped talking to my phones.
Asterisk was still working as I got 2 voicemails. I have TDM analog
cards for incoming calls.
Anyway my cisco phones had X's
On Thu, 2006-10-12 at 10:20 -0400, Jay R. Ashworth wrote:
On Wed, Oct 11, 2006 at 12:07:50PM -0400, Bob Chiodini wrote:
We had a power failure that took down the internet connection and local
DNS server. My local Cisco phones could not register (IP addresses are
hard-coded) and, because
Tom,
The uniden TRU446 and the CLX465 both are supposed to detect stutter
dial tone (SDT) from the phone company and light the MWI. When used
with asterisk the SPA3000 can generate SDT. I'm not sure it can do so
on its own. I gave up on the SPA 3000 due to echo problems.
an external MWI device or I'm
going to have to replace the Uniden phones.
On 10/14/06, *Tom Lynn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
My uniden phone is the TRU8885-3HS.
On 10/14/06, *Bob Chiodini* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Tom
On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote:
Hi people,
When you use a TDM400p with 4FXS i know i need to connect a 12V
connector to power the FXS lines.
Im not good at electric stuff so I ask...If I have a 60W DC to DC
adapter (80W peak) then, how much power will the TDM 400P
, but is not present as a factory default.
So My real concern is power.
On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote:
On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote:
Hi people,
When you use a TDM400p with 4FXS i know i need to connect a 12V
connector to power the FXS
I'm a Bellsouth DSL user in FL too. Here, the filter has a DSL/modem
jack and a POTS jack. So if a phone and modem share the same wall plate
the filter does the split.
I don't think connecting the DSL modem directly the loop is wise.
That's assuming that the filter actually filters something on
when you move. I could put one in an envelope and send it to
your father, or deliver if he's near me. I'm in Brevard county.
Bob...
On Tue, 2006-05-09 at 08:55 -0400, Hadar Pedhazur wrote:
Bob Chiodini wrote:
I'm a Bellsouth DSL user in FL too. Here, the filter has a DSL/modem
jack
Bellsouth gave me a box of filters that have two RJ-11 jacks. One for
the DSL modem and one for a phone. The instructions specified that
every phone be connected to a filter. The DSL modem would then be
connected to the DSL jack along with one of the phones. The modem
should not be connected
Jürgen,
The TAE jack sounds like a great idea. In my house all of the phone and
data cabling is home-run to a punch-down block in a Comm closet. The
single DSL/POTS filter is located there along with the modem router and
a SPA-3000. Other than a nearby lightning strike destroying my filter,
Good Morning,
I've been trying to set up [EMAIL PROTECTED] and thing are going pretty
well. I do have a question: When I *98 into voice mail I hear a
message that says Asterisk mail then short pause then the word
mailbox then a very long pause, then a request for a password. I
believe some
I don't hear a request for my mailbox number. Should it say something
like Enter mailbox number?
Bob...
Avi Miller wrote:
On 25/05/2006, at 8:14 PM, Bob Chiodini wrote:
message that says Asterisk mail then short pause then the word
mailbox then a very long pause, then a request
Avi,
Got it! Thanks. The minimalistic approach :-)
Bob...
On Thu, 2006-05-25 at 21:07 +1000, Avi Miller wrote:
On 25/05/2006, at 8:57 PM, Bob Chiodini wrote:
I don't hear a request for my mailbox number. Should it say
something like Enter mailbox number?
I believe the prompt just
On Thu, 2006-06-01 at 12:30 +0200, Shenen Shenen wrote:
Hi!I've a question:
I've 2 asterisk, I want pull the ethernet wire and then reconnect it
after 5 second, using the VRRP protocol, where must I set the IP for
the connection goes on the second asterisk?
I want this:
I call to asterisk1,
On Fri, 2006-06-02 at 12:01 +0200, Louis-David Mitterrand wrote:
Hello,
At a client site yesterday I installed a dozen GrandStream GXP-2000's
with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX
and phones: network access for users windoze PC's through the phone's
You might put 600 ohm/600 ohm matching transformer to isolate the port
and the amp. Should also maintain loop current if needed.
Bob...
Jerry Jones wrote:
use an fxo interface and 600ohm input on amp
On Jun 11, 2006, at 9:53 AM, Thomas Kenyon wrote:
Doug Lytle wrote:
Thomas Kenyon
Thomas Kenyon wrote:
John Novack wrote:
Bob Chiodini wrote:
You might put 600 ohm/600 ohm matching transformer to isolate the
port and the amp. Should also maintain loop current if needed.
Bob...
FXO ports do not generate loop current, they detect loop current from
Thomas Kenyon wrote:
John Novack wrote:
Bob Chiodini wrote:
You might put 600 ohm/600 ohm matching transformer to isolate the
port and the amp. Should also maintain loop current if needed.
Bob...
FXO ports do not generate loop current, they detect loop current from
Matt wrote:
What on earth is going on with the list?!?! Some of my messages
never make it... then days later I get something like this back:
Final-Recipient: rfc822; asterisk-users@lists.digium.com
Action: failed
Status: 5.0.0
Diagnostic-Code: X-Postfix; mail forwarding loop for
Al Lougher wrote:
Hi -
I hope someone out there can help. I've just built a new asterisk
server running [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 2.7. and I'm
having real difficulty setting up my cable modem for the internet
connection. I have 1xcable modem and 1xnetgear router and 1xPCI
Al Lougher wrote:
Hi -
This is the first Linux server I have ever built with an installation
of [EMAIL PROTECTED] 2.7 mailto:[EMAIL PROTECTED]. For development I
have been running on VMWare on an XP box and sustained no crashes or
reboots. After moving Asterisk to it's own server I am
Francisco Gonzalez Canales wrote:
I forgot to mention the firmware: P0S3-08-2-00
On 7/12/06 4:53 PM, Francisco Gonzalez Canales
[EMAIL PROTECTED] wrote:
Hello,
For some reason it seems like the phone is not getting the file dialplan.xml
file loaded from the TFTP server.
Dialplan.xml
On Thu, 2006-11-02 at 11:32 +0200, Tzafrir Cohen wrote:
On Thu, Nov 02, 2006 at 10:10:01AM +0100, Florian Hars wrote:
Zaptel installs an /etc/modprobe.d/zaptel and an
/etc/{defaults,sysconfig}/zaptel that list the modules in a different
order, so If you happen to have a TDM2400P and a
Switching is what you want.
NAT is Network Address Translation that allows the router to map IP
addresses between router interfaces.
You may wish to verify that all of the ports on your network, if
automatically negotiated, did what you want. Probably, 100Mb,
Full-Duplex. If not then force
Tzafrir Cohen wrote:
On Thu, Nov 02, 2006 at 06:34:03AM -0500, Bob Chiodini wrote:
For Redhat, Fedora, CentOS and other derivatives:
You can play tricks in /etc/modprobe.conf using the install directive.
The man page for modprobe.conf gives an example.
This is not the proper place
I'm in the US and had bad echo problems with the SPA3K and the latest
firmware. I was under the impression that the echo was due to my long
cable run to the CO ~15000'. Changing the impedance (900 ohms) would
help for a while, but after a few days the echo came back. If I rebooted
the SPA3K
I forgot about the levels. One thing that really helps is lowering the
to PSTN gain. Unfortunately, you may find that to reduce echo to a
tolerable level you will need to reduce the gain so low that the called
party may have a hard time hearing you. I think I dropped the gain
values by 4 dB.
On Mon, 2006-11-06 at 21:31 +0800, William Kenworthy wrote:
Is it possible have multiple concurrent ip numbers for a single
extension? How?
I am using a laptop that I move around various local and remote
networks so the IP numbers it uses varies. As I am on extension '205',
I want be
There is an option on the list server membership configuration screen
that will disable receiving your own posts to the list. Maybe the OP
accidentally disabled this feature.
Bob...
On Wed, 2006-11-08 at 06:09 +0200, Dovid B wrote:
I have seen this mainly with gmail. the logic is why do you
Christian,
Could you be out of disk space? What is the output of df -k and mount.
Also does /root/zaptel-1.4.0-beta2/tonezone.h exist? Assuming the
source is at that directory level.
Bob...
Christian wrote:
Hi,
No, I still get that error as before. And I havent installed anything
Christian,
Either mkdir -p /usr/include/zaptel/tonezone or delete the tonezone.h
link then re-run the build. The date stamp on the link does not
correspond with the others the directory. Could be something leftover
from an earlier attempt?
Bob...
Christian wrote:
Hi,
See my answers below.
Dovid B wrote:
I have a client that has a dedicated box. Running asterisk 1.2.10 with
ztdummy on Centos. He is connected via Bell South DSL in the Miami,
Florida area. He has been complaing about voice quality issues. The
person he is calling can hear him fine however he can not has terrible
Kelly,
Could there be a mismatch at the branch switch? Such as ethernet
interfaces operating at half-duplex when the switch is at full-duplex.
This usually manifests itself as dropped packets. I have an older Dell
box that cannot seem to negotiate with a Cisco switch. 50% of the time
it comes
It appears to be up there now. From the header:
## Configuration template for GXP-2000 firmware version 1.1.1.14
; this dial plan may be set to
“none”. This case also belongs to call type #7 and the voice path is (1)
(2) (4) (6) (7).
Of course, I'm having a lot of trouble reading this complex manual g
Larry
Bob Chiodini wrote:
Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in:
http
Eric,
It looks like the definition for PTHREAD_MUTEX_RECURSIVE is within an
#ifdef __USE_UNIX98 (on Fedora Core 6, anyway). You could try defining
it within the Makefile. Similar to the _GNU_SOURCE definition in the
app_cepstral.so: app_cepstral.c stanza.
Bob...
On Wed, 2006-11-29 at 13:38
Giorgio,
You could set up a caching name server in your local network, use it as
your primary DNS server and your ISP's as a secondary. This would cache
your ITSP's address(es) locally limiting your reliance on your ISP.
Bob...
On Wed, 2006-12-06 at 10:43 +0100, Giorgio Incantalupo wrote:
Hi,
Doug,
The Uniden CLX465 supports stutter dial tone (SDT) and provides a MWI.
Might be overkill since it is an answering machine as well. There are a
few others. Google for stutter dial tone or phone company compatible
voice mail. The SPA3K can produce SDT. The Budgetone 102 also has an
MWI.
Gustavo,
Take a look at this thread
http://lists.digium.com/pipermail/asterisk-users/2006-October/169627.html
Presumably the supplemental 12v supply is for ringing voltage.
I did not see anything on Digium's support pages about the card itself.
Maybe a call to tech support may help.
Bob...
It looks to me that if the test clause is false then
ast_gethostbyname is called. Presumably not needed when NAT is enabled.
Bob...
je . wrote:
In chan_sip.c, line 5876 (Asterisk-1.2.13), the
function parse_ok_contact returns whether the host
that requested an invite is a valid or
then ast_gethostbyname will be run - is
that correct? In this case, why the distinction
between a NATted and non_NATted implementation?
--- Bob Chiodini [EMAIL PROTECTED] wrote:
It looks to me that if the test clause is false then
ast_gethostbyname is called. Presumably not needed
when NAT
The free version 1.31 has all 16 keys on the keypad.
Bob...
Al Bochter wrote:
Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF
tones
Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *
The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C
Doug,
Thanks for the info. I'm glad it works.
One question: Is there some sort of one-button way to dial in to your
voicemail? It seems I read something about it, when I was doing similar
research? I think it was the Uniden CLX-465, which claims support of
Phone Company voicemail. I
Kenneth Padgett wrote:
I'm working from the docs here:
http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk
and getting an error doing the ./configure on the iksemel module:
checking for getaddrinfo... yes
./configure: line 20399: syntax error near unexpected token `,'
./configure:
Kenneth Padgett wrote:
Bob,
It looks like the gnutls development package is called gnutls-devel:
'yum install gnutls-devel' should get the package installed.
Yah, I thought that would be it. I have that installed, as well as
gnutls. (I basically installed both packages you can find with
Facundo Barrera - GMail wrote:
Hi list:
This is my first post and first off all i want to wish a good
year for everone! well my problem is; i already installed asterisk on
a server and created a channel and a couple of extensions, all seems
to work just fine, y can make calls and receive
Isn't that what
externhost=sip.server.com.ar my server name on the internet
localnet=192.168.5.0/255.255.0.0 my LAN
is supposed to do?
Bob...
Rudolf Ladyzhenskii wrote:
NAT changes address of the packet, but does not go inside of the SIP
packet itself. And SIP packet contains address as
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