if after you tried both straight through crossover cables and
it still give you RED alarm. just tell them you can't get any
clocking signal. they'll probably send someone on site and test
the line.
Yes, I tried all sorts of cables and ended up getting the local contact
to complain to NETCOM.
On Thu, Jul 31, 2008 at 12:31 PM, Uros Djokic [EMAIL PROTECTED] wrote:
Hi,
Ensure that in file indications.conf you have
[general]
contry=cn ; not usa ! or if you are in Australia shortcut for Australia
Regards,
Uros
--
Use Free Software http://www.fsf.org/
Hi,
Ensure that in file indications.conf you have
[general]
contry=cn ; not usa !
Regards,
Uros
--
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
2) To copy program
3) To modify source code
4) To
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Lee, John (Sydney)
Sent: Thursday, July 31, 2008 3:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie in China: Red alaram in
Zaptel for E1
Sounds like you're making progress. I would try the above span
definition without the crc4. That might do the trick.
Thanks Brad.
I already tried it without crc4 but it makes no difference.
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Ensure that in file indications.conf you have
[general]
country=cn ; not usa ! or if you are in Australia shortcut for Australia
Uros, that was a good reminder. However, I don't think it is related to this
problem.
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Make experiment.Make loopback Rj-45. (wire 1 from pin 1 to pin 4 wire 2 from
pin 2 to pin 5). Then put it in card and if card is OK you should see green
led.You should also see dozens of ALARMS notices or warnings on asterisk
CLI.
Also check pinout http://www.goonda.org/archive/docs/pinout.html
Hi,
If I set maxexpirey=60 in sip.conf and also set a registration timeout=60 on
client software, doesn't this mean that the SIP user (an ATA connected phone)
should be forced to re-register every minute?
If I look at the CLI when the SIP user registers I do see a statement regarding
a 60
Hi guys,
What's the best way to setup a ring group that contains 6 extensions so that
when a call comes in there starts a 30 second timer and the first available
device is rang instead of ringing all extensions at the same time?
What I want it to do is cycle through the extensions and have the
On Thu, Jul 31, 2008 at 05:36:14PM +1000, Lee, John (Sydney) wrote:
Yes, I tried all sorts of cables and ended up getting the local contact
to complain to NETCOM. An engineer came and swapped the Fast Ethernet
to E1 converter.
Hmmm.
Whose side is Fast Ethernet, and whose side is E1?
Are you
Yes, this is really a spam. Yes, it came through the list, not direct
to you as a forgery. It's shown up on several of my other mailing
lists this week, as well, including, ironically, MailScanner's.
People are chasing it.
If you're not the list admin, do everyone a favor, and don't burn up
Why don't you just call the Dial application for each user, one after
another ??
The ones that are busy will just go through. So, on the next priority,
you dial another one.
Tom Moore wrote:
Hi guys,
What's the best way to setup a ring group that contains 6 extensions so that
when a call
J.M. wrote:
I've followed the instructions here
(http://www.voip-info.org/wiki-Asterisk+RealTime) and other places,
however, Asterisk still reads information from the .conf files. How
can I get Asterisk to read from the database and not from the .conf files?
I realize the information
Hi all
I have been looking at my asterisk CDR in the mysql database and
some channel names are set to **Unknown** string.
When I look at the code, everybody when calling ast_channel_alloc set a
channel format
like SIP/%s or Zap/%s
Only app_voicemail.c doesn't when sending emails and I don't use
Sounds more like a hunt group than a ring group.
Bruce Komito
WPTI Telecom
(775) 236-5815
On Thu, 31 Jul 2008, Ruddy G. wrote:
Why don't you just call the Dial application for each user, one after
another ??
The ones that are busy will just go through. So, on the next priority,
you dial
This works only half way.
This gives the ring function I want, but doesn't take in to account the 30
sec timer to send to voicemail if the line is not answered.
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ruddy G.
Sent: Thursday, July 31, 2008
When I do a reload in the Asterisk CLI I get a long list Unregistered
indication country lines during the parsing of the features.conf file.
Then, when parsing the indications.conf file, they seem to all get
re-registered (lines saying Registered indication country are displayed).
What do these
This release primarily focuses on security.
A number of problems involving SQL injection
and XSS were identified and reported by Jean-Michel
Besnard.
Jean-Michel was kind enough to help with the testing
as each vulnerability was addressed.
The new release is available in the downloads section
Hi All,
i have asterisk with 9 SIP accounts on it.
i was wondering if theres a way to setup asterisk, to send the amount of
minutes each SIP account have spent incoming as well as outgoing and if
possible the number it called!
any advice?!
any help would truly be appreciated..!
thanks in
Astricon is only 54 days away! If you're not
booked, please take a moment to register for the
conference, get your hotel room, and get your
plane tickets before things fill up and/or get
expensive. This is a great opportunity to meet
other developers, users, and members of the
Asterisk
You can check asterisk CDR (call detail records).
You should have a csv file in /var/log/asterisk/cdr-csv/Master.csv
You can also configure it to write the CDR in a database
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
Then you can just write a script that will look at your database and
send
If I buy two AA50s can I set them up so that everything runs through the
first one, but the second one will take over if the first one goes down?
I can see the extensions recovering, because they use ethernet, but
what about the FSO lines? Is there a way they can be spliced to both
AA50s so
I'm presently working on an office move and evaluation of telecommunications
services needed at the new location. I'm presently wrastling with an issue
related to portability and geography between landline carriers. Presently
certain people within the organization are hopelessly in love with our
I've been looking at various solutions for getting FXS and FXO lines in and
out of asterisk. one solution is using TDM-400 cards. Another solution is
using the grandstream GXW400x and GXW410x gateways. Cost per port seems
lower on the gateways and no pci slot is required. Why would one choose
Eric Fort wrote:
I'm presently working on an office move and evaluation of
telecommunications services needed at the new location. I'm presently
wrastling with an issue related to portability and geography between
landline carriers. Presently certain people within the organization are
Might not be lower in cost but when you take into account the cost of
the server it would be - how about checking out the Vdex-40 appliance if
you need 4 pots lines or less.
http://www.taa.com/products-vdex-40.html
Cheers,
Dean
From: [EMAIL PROTECTED]
There is 2 possibilities to do failover with asterisk:
First use openser with failover, but still you need to switch cables fxo fxs
pri bri
Second is simplier and i would choose this one for smallmedium
installations where T1/E1 is not needed:
It consists of externaising of all fxs fxo pri bri
This is true.
Probably is a hunt group.
Different systems use different terminology for the same thing sometimes.
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito
Sent: Thursday, July 31, 2008 11:19 AM
To: Ruddy G.
Cc: Asterisk Users
With use of tdm you can get 0 lag when using faxes modems over FXO FXS
With use of fxo fx gateways it is easyer to build redundancy with
heartbeat for example.
2008/7/31 Dean Collins [EMAIL PROTECTED]
Might not be lower in cost but when you take into account the cost of the
server it would
On Thu, Jul 31, 2008 at 02:10:34PM -0400, Dave Welsh wrote:
If I buy two AA50s can I set them up so that everything runs through the
first one, but the second one will take over if the first one goes down?
I can see the extensions recovering, because they use ethernet, but
what about the
you have this option on major phones also, try that.
2008/7/31 Vieri [EMAIL PROTECTED]
Hi,
If I set maxexpirey=60 in sip.conf and also set a registration timeout=60
on client software, doesn't this mean that the SIP user (an ATA connected
phone) should be forced to re-register every minute?
i saw that billing iface somewhere else, maybe i am wrong...
2008/7/30 Mindaugas Kezys [EMAIL PROTECTED]
Hello,
Based on our own and our clients' experience we compiled short manual: How
and to whom sell VoIP
Hope it can be useful to some of you also.
You can download it from our site:
I'm considering getting a Panasonic video door phone system (VL-GM301A)
which can interface with a PBX and would like to connect it to my
Asterisk box with an analog FXS port. Of course the Panasonic
documentation only talks about hooking it up to a Panasonic PBX which
only talks about using
From my research, it seems that for FXOs you can use a siple RJ11
splitter. A special splitter that gives priority to the backup split
is preferred. These will sometimes be used for old answering machines
where the handset can overuse the answering machine if it's picked up.
For the server
On Thu, Jul 31, 2008 at 01:20:00PM -0700, Eric Fort wrote:
I've been looking at various solutions for getting FXS and FXO
lines in and out of asterisk. one solution is using TDM-400 cards.
Another solution is using the grandstream GXW400x and GXW410x
gateways. Cost per port seems
Happy August.
After two fiascos, let's try this again. I'm not positive John Todd
will be available, so we will play it by ear. If he is, we can talk
about Astricon news and Asterisk User Groups as planned. If John can't
make it, we'll talk about anything anyone wants to discuss. There is
plenty
Hi there,
Is anyone using a headset with one of these phones? If so, can you
recommend any?
Thanks
Simon
___
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now:
Tom Moore wrote:
This works only half way.
This gives the ring function I want, but doesn't take in to account the 30
sec timer to send to voicemail if the line is not answered.
What you are looking for is a 'queue' in Asterisk terminology. These
already exist and can be built and managed
Plantronics.
PaulH
Simon wrote:
Hi there,
Is anyone using a headset with one of these phones? If so, can you
recommend any?
Thanks
Simon
___
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AstriCon 2008 - September
Hi I am new to asterisk and to the AMI.
I have been automating calls using the AMI's originate, this has been
working fine for me. I have been calling from one end point registered
with the asterisk to another endpoint registered with the asterisk.
Now I want to be able to call from a known
So any 2.5 headset will work with the SPA922?
On Fri, Aug 1, 2008 at 12:23 PM, Paul Hales [EMAIL PROTECTED] wrote:
Plantronics.
PaulH
Simon wrote:
Hi there,
Is anyone using a headset with one of these phones? If so, can you
recommend any?
Thanks
Simon
That's a good question - the plantronics are available with
interchangeable ends - which makes them easy to move between different
phones.
PaulH
Simon wrote:
So any 2.5 headset will work with the SPA922?
On Fri, Aug 1, 2008 at 12:23 PM, Paul Hales [EMAIL PROTECTED] wrote:
Plantronics.
C F,
Does the 2nd port of the ATA with 2 FXS ports just work like a
'pass-through' that is connected to the DTMF Relay? Or am I totally
off track?
Any ATA's with 2 FXS ports that you can recommend?
Thanks,
Julian
On Thu, Jul 24, 2008 at 3:27 AM, C F [EMAIL PROTECTED] wrote:
leave the
Are there are any xmpp developers on this list?
I might have a small consulting project to build an XMPP chat
application/(or even better alter off the shelf application with desired
customizations)
Email me for details.
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357
Hi,
I have this weird problem i cant explain.
i have two asterisk, i'm using realtime table for my sip/user accounts.
my database is on a mysql cluster.
my prob is if i register on phone on asterisk 1 it is ok, but on second
asterisk it can't,
Registration from '122144 sip:[EMAIL
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