only condition would be that you do not use it for a
commercial use, i.e. you don't try to sell a t.38 module for asterisk.
If you want to retain any control of what it is used for, you better
re-register it. Once it expires and some one else gets it, you have no say
in the matter.
John
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
I'd guess the battery on your motherboard has died so it is
going back
to 1970 at boottime.
That would be 1980, right?
Only if this was 2018...
No. Looks like some signed/unsigned int error somewhere.
The lost bios makes more
But each time the pound key is pressed, file dir-intro.gsm
will always be played. Can any one tell me the reason?
Because you have enabled the directory in your IVR. Uncheck the Enable
Directory box on the IVR screen and you can then use the octothorpe.
By the way,if I want to modify the
Why everybody charge so much for this information,
why this information could not be free ?
Probably because Telecordia makes a decent amount of money from selling it.
If they gave it away that revenue stream would dry up.
John
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Does anyone make an interface card that can integrate with
the digital input of the Meridian. Not the optimal solution,
but it allows for the current infrastructure to be retained.
By digital input do you mean a T1 interface? If so then yes several T1
interfaces are available. However I
Well, I am not sure what is needed to interface between the
two. I hoped there was something you could use and from the
Joseph,
Now I'm pretty sure we are not talking about the same things. Let me see if
I have the correct picture in my head. I now think you have a Norstar in one
office and
-Original Message-
Subject: [asterisk-users] Diagnosing dropped calls...
I have a system that is driving me nuts. My customer
is running Asterisk 1.4.20.1 on a CentOS 5.2 server. It is a
purely SIP and IAX2 service with no cards installed and it
uses ztdummy from Zaptel
Try dropping the IAX2 and only use SIP. Don't ask why?
Well in our case we were NOT using IAX at all. Strictly SIP.
You could be hitting an overloaded router or whatever along
the way, 10mbs fiber does not mean low latency or lost packets.
So true, hence the reason I suggested using mtr to
kill
it.
John Faubion
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need auto answer or how else would the audio come out of
the speakers?
One of the lowest cost solutions I've used for this is a Grandstream BT-200.
You can buy the phone for around $50, it supports auto answer, and it has a
miniature audio jack already on it. Just run the output to an
As far as a license is concerned, we do not ship with any
codecs that require licensing (we support them) and when
someone purchases an ISPBX PBX system, the license for using
What Kristian was asking is, what license does the software you have written
use? Is it GPL? Seeing that many of us
Forwarding isn't on. If I call from my cell: fax machine.
Call from a land line: call gets through. Call from Skype:
fax machine again!
Ok now that is bizarre. Three different sources and two different
destinations. Looks like the only common point would be BV. Sounds like your
going to
cell phone. When I do, I get a fax macine. Debugging SIP
shows NO call activity what so ever.
Make sure you don't have it forwarded to another number at BV.
John
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are plenty of phones on the market which do SIP now - most
modern Nokias do. I use an E90 Communicator, but the E95 is
popular too, so I'm experimenting with using my mobile as my
one phone, via Wi-Fi/SIP when I'm in the home/office and
Out of curiosity, how do these phones handle the
I reboot every evening :) Drew, what's the uptime on your
asterisk process on that box that's been up for 193 days?
I too restart the asterisk process every night as part of the cron process.
Many people here seem to be under the impression that restarting the
application every day is a bad
Although this is a users list, I think it is more of a list for
Asterisk resellers. I'd be interested in how many of you are simply
using Asterisk as your phone system and NOT selling your services or
an Asterisk based solution?
I actually work as a software engineer for a big telecom
For such a small system there is no earthly reason for it to
be 10 percent of that, even on a 5 year lease.
I know that EVERYTHING is big in Texas, but that is nothing
more than highway robbery.
I fully agreed, that's why we built her an Asterisk based system. Splitting
this up they wanted
when you look at the iPhone with all its amazing features for less than
$500.00 it just doesn't make sense. Am I the only one that thinks this?
Remember that the service providers such as ATT, Cingular, Sprint, Verizon
and so forth, subsidize the cost of the phones because they make it up
and the music industry.
John Faubion
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I've had some serious issues with Teliax as of late with their new
Denver server. DTMF issues, IAX2 connection issues, and major
latency issues. They are blaming it on 1.2 vs 1.4 asterisk compatibility
issues. I have had zero problems with their old servers.
Interesting... I've got several
A while back i had asked about possible replacements for snom 360 phones
that were breaking and causing
issues and we all discussed the problems we had with the 360s and some
suggestions were made but the
new polycom phones had just hit the market and not many people were able to
comment on them.
Adam Moffett wrote:
So you want a device that will answer a SIP call, and play the
audio out
to a speaker?
You're looking to build a PA system then?
We achieved this using a Grandstream Budgetone configured to
auto-answer, and just soldered a pair of wires across its speaker
Neither
suggestions to the cards of question. Is your issue with his remarks somehow
related to your consultation business?
John Faubion
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No one else is seeing this issue ?
I have the same issue but I haven't put much effort into solving it yet. Too
many other issues seem to get in the way.
John
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Well can you offer some explanation why T.38 faxing worked for months
and then one day stopped working?
Generally it is because some one or some process did one or more of the
following:
1. Updated firmware on the ATA
2. Updated software on the server
3. Changed a configuration setting
4. Let
The releases are available for immediate download from
http://downloads.digium.com/.
Could someone make sure the files are actually available BEFORE sending
these out?
John
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Could someone make sure the files are actually available BEFORE sending
these out?
I apologize for the way that sounds. It certainly sounded a lot more
tongue-in-cheek in my head.
John
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I guess /tmp can live in RAM, but what about eg. recording ten-twenty
WAV files to /var a day, and logs into /var/log? Do I have to worry
about the card wearing out in six months?
This is nothing really. Just make sure your using an industrial compact
flash card. These support 1-2 million
Is the PCI slot large enough for full height, half length PCI boards ?
Yes.
Has you heard of a PCI Express version ?
No but the way chipsets are coming down in price, I would imagine someone
will have it soon.
John
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Thanks for the tip. It seems like they no longer manufacture them:
http://www.neoware.com/products/hardware/
No, but the Neoware e140 has a PCI expansion slot, is expandable to 1GB RAM,
and still has room inside the case for a hard drive. It is available without
Win XPe starting at $339 new.
I'd rather use a PCI card to connect * to the POTS, and a hard-disk
instead of a CF card. Do you know of a similar, small form-factor
motherboard + case that would fit the bill?
Many of the thin clients fit the bill nicely. I've been using MaxSpeed
MaxTerm clients lately. Mainly because I
I`m using several GXP2020 phones with newest Firmware 1.1.4.18.
I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22
and have eliminated that.
Asterisk Version: 1.4.11.
Me too. Still testing 1.4.13 on a non-production system.
I use on every phone the 1 as local
Am I the *ONLY* one that has this issue?
John Faubion
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Faubion
Sent: Thursday, November 01, 2007 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RTP
to correct this? I'm using SIPConnect from CBeyond and this
appears on incoming calls. I haven't had any complaints about voice quality
and I haven't seen any dropped calls. Should I be concerned?
Thanks,
John Faubion
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Let's suppose that my sip extension 3000 want to call to (302).123.3211
I need a rule in extensions.conf to match with this number, right ?
Let me see if I have this correct. You want to use the
provider-302333-3000 for any call going out from 3000 and
provider-30-3001 for any call going
has anyone actually been satisfied with the performance of these
powerline signalling devices ?
Universal Powerline Bus is a vast improvement over the original X10. I
believe the X10 devices used a 4V signal during the zero crossing point of
the AC voltage to transmit 1 bit. Needless to say
Does anyone know of such a device that I can use over a network? It would
be a pain to run a USB cable. I am thinking of devices that are like:
I think your missing the key feature of these devices, UPB/X10. UPB and X10
are communication protocols that runs across the electrical wiring in the
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in
a business graded installation (with really traffic on not 3
calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4
I have an installation right now in a real estate/mortgage company office
with 36
The obvious alternative is to use the extension as the sip UID:
Use the extension as the UID and add the mac address as a comment. Like so:
[123]
; Joe Smith
;mac=000E08DA0409
secret = blahblah
... and so on and so forth
This will give the best of both worlds. The mac is readily available and
I got my G729 licenses installed.I can make calls out and receive
Make sure you add g729 to the voicemail config as well.
John
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I can't even hear the password prompts.
Ahh... have you loaded the G729 sounds? Are you getting errors in the logs?
John
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To prevent further missunderstanding please do not refer the SI-120 as a
snom
phone. If you need support please contact snom India.
Tim,
If it is sold by snom India, and one is to contact snom India, I can
certainly see how one could infer that it is indeed a snom phone.
John
I can only pointing out this issue, trying to prevent bad user experience
with these phones.
I'm very very unhappy about the situation but it's out ouf my reach to
change this.
Yes, and I certainly understand how difficult it can be to get upper
management to understand that their cost
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
I've seen several people mention it taking a few days to send messages. I've
usually seen mine in a few minutes. We'll see about this one... sent July
4th
Is it just me? After the mail list server upgrade, the average delivery
time for messages to the users list is between 4 and 5 days. The Dev
I've seen several people mention it taking a few days to send messages.
I've
usually seen mine in a few minutes. We'll see about this one... sent July
4th
They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ?
Yes, and just to complicate matters further, they are probably asking about
the NT-1 or NT-2 which is the Network Termination type. NI1/NI2 usually
refers to the National ISDN phase, for which the difference has generally
We do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
The issue really isn't whether you have the ability to make toll calls on
your line. The
When making an outbound call, the outbound peer return a 301 forwarded
with URI to other
domain, but asterisk think it's a local domain and try to look it up from
extension.conf.
What phones are you using? This sounds a lot like a problem, I have using
Grandstream phones.
John
What is a god Windows application to read core dump files?
Microsoft jokes aside, I would seriously doubt there could be a good Windows
application for analyzing core dumps. Due to the OS specific nature of core
dumps, the need to have the source files, debugger and more, would make it
difficult.
by the way selling does not depend on the amount of lines you have and
we are very productive trust me
True, very true. There are lots of very productive sales people that don't
need a phone at all. From the paper boy to car dealers, lots of sales don't
require many phone lines. Of course at
Any recommendations on an economical layer 3 switch?
I've been quite happy with the Netgear FS728TP ProSafe switches. These are
24 port 10/100 switches with 4 GigE ports and PoE. They also support 1000-SX
GBIC through an optional module. The total PoE budget for all 24 port is 195
watts. We
Polycom Phones
1. New call
2. Press 9 access outside line
3. Dial Cell Number
4. Transfer the call that way.
Once you initiate a new call you will tie up the second line. Your asterisk
box will now be bridging the two lines. The lines will stay tied up until
the salesman drops the call.
One
If you really need layer3 support, I would steer clear of the Netgear.
I've had a lot of problems
with them, and the support was disappointing.
What model did you use? I've been very happy with the FS728TP as I mentioned
earlier. I haven't had any problems so far. Granted I haven't had to call
I would also like to know if Asterisk can be setup to automatically re
start if there is a core dump.
Sure! You should already have the required script. Just run it from
safe_asterisk. Here is a link with more info:
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs
Dude you got to be freaking kidding me - are you really sending this
email to everyone who posts on the Asterisk list?
No, most likely he has an autoreply/vacation/out-of-office message enabled.
I would expect us to get more of them. Just be thankful he is on digest
mode!
John Faubion
as the salesman's cell phone provider and
your mobile to mobile minutes can be free. If you have more than a couple
salesmen, this route will likely entail a multi-port gateway but the idea is
still the same.
As far as the right way, that depends on way to many factors tat you
haven't addressed.
John
Anyone get it working on 1.4. Checked out their
website no updates for some time now...
Sorry for the late post. I'm tying to catch up on my email. I have three
systems using it. One with 1.4.3 and the other two on 1.4.5 if you still
need help.
John
Carsten Bock wrote:
See the 1.4.5 Changelog:
(http://ftp.digium.com/pub/asterisk/ChangeLog-1.4.5)
2007-06-07 19:46 + [r68196] Olle Johansson [EMAIL PROTECTED]
* channels/chan_features.c: Disable chan_features by default in
menuselect
Well isn't THAT enlightening?!? Anyone
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