You've hit the nail on the head with the crux of the pain I went
through. Finding stuff that was broke that I didn't realise was broke
until someone bothered to tell me about it. I'm sure everyone is
familiar with just how often users report problems caused by themselves,
but don't report stuff
Speaking of attacks that aren't fair. Trixbox != FreePBX. They're
completely separate products.
Tilghman Lesher wrote:
FreePBX seems to be the most logical choice to me.
Which is being leveraged to take away business to anyone who has not sworn
allegiance to Fonality. Sorry, couldn't
Unless your provider provides a T.38 gateway, fax over SIP is pretty
much guaranteed to be unusable. Often you can get away with it over a
LAN using G711a or G711u, but any of the lower bandwidth codecs /won't/
be able to properly handle fax calls.
Whilst I haven't used it myself, I believe
Well that answers that question. I see that t38modem provides an H232
modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact
that it requires a kernel recompile on most newer distros.)
Steve Underwood wrote:
Rob Hillis wrote:
Last time I heard IAXModem didn't support T.38
Then I suggest you prepare yourself for a lot of pain. Fax over the
'net without T.38 is almost guaranteed to not work.
Al lists wrote:
I'm not looking at T.38 , at this time its terminating a SIP trunk
with multiple DID's for fax.
I'm using this configuration with linksys PAP ATA and
I'd say consider yourself very lucky. I know I did some testing here
some time ago with faxing over VoIP.
* One extension to another over G711a with both extensions on the
same LAN - worked 95% of the time
* One extension on my Asterisk server to an Extension on a friend's
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Hillis
*Sent:* Tuesday, January 01, 2008 4:13 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk 1.4 Fax
I'd say consider yourself very
Apparently not. I'm sure as heck not going to get involved in this
argument again! :)
Bill Hackensack wrote:
Do people even read the mail list anymore, or do they just land on
this planet, subscribe to the list, and ask the same questions that's
been asked over and over and over and over and
The reason that IAX2 is considered good for NAT issues is that it uses
only one port for both control messages and voice traffic as opposed to
SIP that uses a predictable port for control messages and an
unpredictable one for voice/video traffic.
If both servers are behind NAT servers, you /will/
) wont need any port forwarding.
Tim.
On 2 Jan 2008, at 10:18, Rob Hillis wrote:
The reason that IAX2 is considered good for NAT issues is that it
uses only one port for both control messages and voice traffic as
opposed to SIP that uses a predictable port for control messages
If you post console logs as suggested, someone might be able to offer
further suggestions.
William Herrera wrote:
Sip show peers will show the phone connected.
The phone display shows connected with IP, date and time and ... network
Icon not blinking.
If I access the phone through the web gui
:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Hillis
*Sent:* Saturday, January 05, 2008 2:36 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] iP0020 Phone busy signal all the time.
If you post console logs as suggested
Google is your friend.
http://www.google.com.au/search?hl=enclient=firefox-arls=org.mozilla%3Aen-US%3Aofficialhs=jR6q=asterisk+iax+two+serversbtnG=Searchmeta=
Shane D wrote:
Okay, here's the dal.
Me and my friend both have asterisk boxes. I want to be able to type
extension 27 on my end and
I would suspect that your hardware is the cause of your problems.
Running a production PBX system on a discarded desktop system is a
/really/ bad idea.
I would seriously consider an upgrade to your hardware.
Ira wrote:
At 12:34 PM 1/18/2008, you wrote:
Although for some of us, or at
What you run it on is very much a function of how reliable you want the
system to be. The better the hardware, the more reliable it will be.
If you're running in a business environment, then I wouldn't recommend
anything less than server grade - even if it's low end server grade.
The
is in putting
the older hardware into full production.
Tzafrir Cohen wrote:
On Sat, Jan 19, 2008 at 06:21:15PM +1100, Rob Hillis wrote:
I would suspect that your hardware is the cause of your problems.
Running a production PBX system on a discarded desktop system is a
/really/ bad idea
PC's age and when they age, things tend to go wrong, particularly when
you upgrade software. Unusual crashes are usually the first sign that
something is going wrong.
To me, it sounds like you've put the money into many of the right areas
- segregating your voice and data networks, going with
Not the first time I've seen something like this happen. If you read
what I said, I wasn't saying that this /was/ what was happening with his
hardware, merely that it's the first sign.
Tzafrir Cohen wrote:
On Sun, Jan 20, 2008 at 06:33:31PM +1100, Rob Hillis wrote:
PC's age and when
Pretty easy actually.
- zaptel.conf
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
zapata.conf --
usecallerid=yes
hidecallerid=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
relaxdtmf=yes
rxgain=-3.0
Yes, but I already knew that. :)
Paul Hales wrote:
I love writing dialplan, using vi.
Does that make me weird?
PaulH
On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote:
Hi, all. I've done some Asterisk recelling, but recently got roped into a
Sr. SysAdmin position. Our PBX
Asterisk really comes into it's own with cute scripts that can do almost
anything with ridiculous ease. One of the things I've done with a
number of Asterisk machines is to put in a script that downloads the
latest weather forecast and reads it back to you using a TTS engine.
Ken D'Ambrosio
You know full well I'm not related to you - I just work with you. :)
Paul Hales wrote:
With comments like that people are going to think that we aren't
related.
PaulH
On Thu, 2008-01-24 at 16:46 +1100, Rob Hillis wrote:
Yes, but I already knew that. :)
Paul Hales wrote:
I
Hi Eric,
You may want to contact me off-list - the company I work for offers a
product which aims to be a zero configuration service for Asterisk. The
Linksys 942 and 962 phones /are/ supported.
Erick Perez wrote:
Hi there,
We have plans to install an office (not call center) with the
I doubt that chan_oss/chan_alsa directly support echo cancelling.
However depending on exactly how you are using the inputs and outputs on
a sound card, you could very well need echo cancellation.
90% of the time, echo is generated at the junction between a channel
that separates received and
Likewise here. The company I work for sells duo core boxes (though
mostly /with/ E1 cards) and we have no issue with timing.
Chris Bagnall wrote:
My question is if anyone else have seen this and if anyone has a
possible solution?
Nearly all of the boxes we've built over the last few
Leave your zaptel config as it is. In zapata.conf, configure your
channel group as being channels 1-15,16-20.
It does no harm at all to configure all the PRI channels in
zaptel.conf. Since Asterisk refers to zapata.conf when it comes to
actual utilisation of the channels, that's where you
According to voip-info, the syntax for the VoiceMail command is as
follows...
VoiceMail([/flags/]/[EMAIL PROTECTED][EMAIL PROTECTED]boxnumber3]/)
If you check the syntax for the VoiceMail command, it indicates that the
mailbox parameter is /not/ optional, so I'm surprised this works at
all.
Why are you specifying the password and server IP in the dial string
when it's included in sip.conf? It's unnecessary.
I believe that Dial(SIP/gs102/1234) will achieve what you want.
ast guy wrote:
Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial
in Database. so will that work in this scenario ?
-ag
On Feb 10, 2008 11:55 AM, Rob Hillis [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Why are you specifying the password and server IP in the dial
string when it's included in sip.conf? It's unnecessary.
I believe
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi guys,
I've been working on a little dialplan fragment for roaming extensions,
however the customer wants us to set the MWI indicator for the roaming
extension that has just logged in. We're using MySQL realtime, so I've
figured out that
the asterisk commands. You execute asterisk command
via agi not using system command
-ag
On Feb 11, 2008 11:24 AM, Rob Hillis [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi guys,
I've been working on a little dialplan fragment for roaming extensions,
however the customer wants us to set
One thing to keep in mind is that the Grandstream's firmware is
notoriously buggy and unreliable. I've got one GXP2000 here that is on
the 1.1.5.15 firmware, and I wouldn't even consider upgrading other
phones to them. Unfortunately, the quality of the Grandstream firmware
is appalling and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If it is being removed in 1.6, I'm a little concerned since there's no
mention of this when you show the application, nor on voip-info.org.
What application/function is it being replaced by?
Atis Lezdins wrote:
| On 2/13/08, Rob Hillis [EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Atis Lezdins wrote:
| By RealTimeUpdate do you mean func_realtime? It shouldn't care, as
| cache is not implemented in realtime level, but higher (chan_sip).
|
| Are you sure you need sip show XXX load. If you sip prune peer
| data, it should be
ugly to me.
Frankly, I'm surprised that RealTimeUpdate doesn't contain an option to
flush and reload details, which would negate the need to employ other
hacks to achieve this.
Atis Lezdins wrote:
On 2/12/08, Rob Hillis [EMAIL PROTECTED] wrote:
If this is the only real alternative
it
for removal in 1.8. Half a development cycle isn't a very long time for
a warning that a function will be removed.
Atis Lezdins wrote:
On 2/13/08, Rob Hillis [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If it is being removed in 1.6, I'm a little concerned
Johansson Olle E wrote:
So please rememner that there are a few independent regular Asterisk
developers out there that is not on the Digium payroll and still take
part in decisions about Asterisk.
Point taken.
Over a year is a long time for a warning like this, considering that
As far as I'm aware, only the Aastra 57i with three 560M modules would
come close to your requirements.
The 57i can display up to 5 extensions at one time with a further 15
being available by the use of multiple pages. The 560M modules can
display up to 20 extensions at one time with three
I guess we ought to add ...beyond downgrading the firmware to 2.0.2 to
that. :)
Paul Hales wrote:
We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and
the BLF indicators no longer function.
Has anyone had a similar issue? And a solution?
PaulH
The cards themselves are okay, but the extra level of configuration is a
pain in the proverbial. Zaptel is already double-configured in both
zaptel.conf and zapata.conf (that's not a complaint - I understand the
reason for the separation) but the Sangoma cards require a /third/ level
of
No. That's how we determined it was the phone and (therefore) most
likely the firmware at fault.
After we downgraded the firmware, the phone did correctly pick up it's
hints.
Sigma Networks wrote:
Paul Hales wrote:
We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30)
You missed a slash.
v
exten = s,n(a42),Dial(${${HuntMember}*/*${EXTEN}}${ds})
^
Louwrens Benadé wrote:
Hi Greg
Thanks for the input. Just to make sure I don’t break my system, could
I run the proposed changed
Of the three ATAs I've got (Linksys PAP2-NA, Sipura SPA-2000 and
SPA-3000) the Linksys PAP2-NA is the best of the bunch, even though the
SPA-2000 is supposedly cut from the same mould.
For the most part, you set 'em and forget 'em. Most of the time when I
have a problem with a phone connected to
They have their ups and downs. If you live outside the US, localising
your tones is a pain in the proverbial since you have to define every
tone by frequency combination and intervals, although I guess you do
only need to do it once.
One other shortcoming of the 942 is the lack of any usable
So far I've never run into anything that's even /close/ to the
speakerphone quality of the Polycoms. There's no comparison on the
speakerphone between the Linksys phones and the Polycoms - it's chalk
and cheese, but by the same token that holds true for just about every
other phone too.
Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk
has had passthrough support for T.38 for a while (somewhere in 1.4 it
became available IIRC) but is currently completely incapable of
terminating or encoding a fax call to T.38.
The only real option available at the moment
.38 gateway.
Steve Underwood wrote:
Rob Hillis wrote:
Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk
has had passthrough support for T.38 for a while (somewhere in 1.4 it
became available IIRC) but is currently completely incapable of
terminating or encoding
About the only reason for eliminating SpanDSP is compatibility with the
GPL license. Remember that /any/ feature added to the free version of
Asterisk is going to be added to ABE as well - ergo, the licensing of
any libraries required need to be compatible with a /non/-open source
license.
.
If you use G.729 in the same asterisk as my spandsp library, you are
breaking my licence conditions.
Steve
Rob Hillis wrote:
T.38 is a codec in exactly the same way that GSM or G.729 is a codec,
so yes it /can/ be used at the same time as any other codec - just
that only /one/ codec
Tzafrir Cohen wrote:
I'm also curious as to why you assert that using G.729 in Asterisk
(/not/ ABE) at the same time as a T.38 implementation that relies on
SpanDSP since these are two completely separate plugins that are
installed and acquired separately.
They are not installed
I agree, the latest firmware does make a difference to the speakerphone
quality, however it still doesn't hold a candle to a Polycom. That's
not to say it's /bad/, just that it's only good, not absolutely spectacular.
The sound quality from the Polycoms is in a class of it's own. It's
like
Did you pay attention to the following bit?
--8snip
For simple dialplans first edit features.conf as desired, then put this
into your extensions.conf
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf:
include = parkedcalls
Posting the same question three times at 12 hour intervals will not get
you a faster reply.
Especially not to an original email that was written in November of last
year.
Joel Solanki wrote:
Hi marek,
Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came
oi geli wrote:
Hi,
I am using SNOM phones with Asterisks for few years. They used go occasionaly
NR, and I would reregister them from the phone web interface. But it started
doing frequently for last few months.
Here are SNOM Phone and the firmware version;
snom190-SIP - Version-Code:
Chris Bagnall wrote:
First things first. You are running /very/ old versions of firmware -
particularly on the 300 and 320. Upgrade them. I've been running
7.3.14 for some time without a problem, though it appears that 7.3.23 is
now out.
I concur about upgrading the software, but
Tilghman Lesher wrote:
Regardless of how you think it should work, the poster above described
precisely the way it works. If your end boundary is 12:00, it will evaluate
as true all the way up until 12:01:59. If you don't want that, another poster
has suggested using 11:59, which will work
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]:
Indeed it does. You add contacts and set the softphone number to
extension@server
Leif Neland wrote:
Philipp Kempgen skrev:
Leif Neland schrieb:
Mostly to debug/test BLF, is there a softphone or another app. which can
subscribe to hints on Asterisk?
X-Lite?
Leif Neland wrote:
I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B)
return busy when just one extension is busy.
Forgive me for the question, but /why/ do you want this behaviour?
Isn't the whole point of dialling multiple extensions so that a call has
a greater chance of
On 12/23/09 12:23, Russell Bryant wrote:
Wasn't this scheduled for 1.6.2?
I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
as best I can tell from looking over the source code :-)
Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze.
On 01/10/10 05:03, --[ UxBoD ]-- wrote:
Hi,
I use VoIPTalk as my provider and unsure of a minor issue. When people call
me they get a US ring tone instead of UK. Is this a Asterisk configuration
issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks.
This is almost always
On 01/15/10 17:54, randall wrote:
hi all,
i noticed that a lot of VOIP phones have a double network interface
allowing you to use only 1 LAN cable for both the phone and your
desktop, a really nice feature that can save a lot of cable, but most
are 10/100 connections while i have a
On 01/16/10 04:27, Bruce Nik wrote:
Hi Guys,
Other than than yum repository (which fails when installing freepbx
with it) are there any automated install scripts out there that would
install Asterisk 1.6 or 1.4 onto a CentOS LAMP system?
If the script install FreePBX that would be a BONUS.
On 01/17/10 01:15, Tzafrir Cohen wrote:
Try PBX-in-a-Flash. Undoubtedly it won't do everything you want out of
the box, but I suspect it will do /most/ of what you want out of the box.
But will not let you debug that install script. I tend to distrust
running such a hidden script
I think he's referring to the fact that you seem to be looking to put
together the telephone equivalent of a spam service.
I'd be advising rm -rf / as well.
On 02/06/10 16:19, Thomas Perron wrote:
karl,
does it make you feel good ?
wow. pathetic.
On Fri, Feb 5, 2010 at 11:00 PM, Karl
On 02/15/10 20:00, Randy R wrote:
Olle, this may be a stupid question, but shouldn't a native santitize
function be urgently added to the code base in all versions or change
the dialplan compîler to ignore dangerous characters?
Whilst I agree with this, the unfortunate attitude we seem to
On 02/17/10 03:39, Warren Selby wrote:
I had a customer ask me this question today, and I was surprised to
say I didn't have an exact answer for them. They have a relatively
small support queue for their business (three agents, and rarely more
than one person in line at any given time in the
On 02/17/10 05:01, Steve Howes wrote:
On 16 Feb 2010, at 17:36, Arjan Kroon | Mobillion wrote:
We are using asterisk version 1.0.0.
Wow.
Yeah, that about sums it up. A little googling reveals that Asterisk
1.0 was announced on January 14th, 2005 - over five years ago.
I would
Your best option is likely to be to create a separate context that calls
both numbers, like so...
[agents]
exten = 1,Dial(SIP/0317998975SIP/0317998985)
...then add Local/1...@agents to the queue.
On 03/14/10 00:03, Magnus Benngård wrote:
Hi!
We have alot of users who are having 2 phones, 1
), W:0, C:1, A:0, SL:0.0% within 0s
Members:
Local/1...@agents (dynamic) (Not in use) has taken no calls yet
No Callers
I am completly lost. :(
On Sun, 14 Mar 2010 01:08:53 +1100, Rob Hillis wrote:
Your best option is likely to be to create a separate
Exactly what is the problem you've having with CallerID? Are you not
receiving it, or are you not able to send it? Which carrier are you
using and what make and model card is the line connected to?
For incoming calls on ISDN-10/20/30 lines, no special configuration is
required to receive caller
On 05/21/10 09:07, Leif Madsen wrote:
Danny Nicholas wrote:
If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in
the spirit of your question:
(1) dialplan conversion
(2) loss of functions like Gosub
Can you be more specific about what 1) and 2) mean?
On 06/19/10 15:19, Kamonwat Sookkara wrote:
Dear Asterisk friends,
Please help me to clarify my doubt. After monitor SIP and RTP
traffic with Wireshark. I found that both SIP and RTP traffic between
2 sip clients must be passed through Asterisk.
Is it possible that 2 sip clients
On 08/11/10 18:46, Tino wrote:
Thanks Gareth for your quick reply.
It is the lateset version and i think i need access to Dahdi
interface. Is there any disadvantages other than this.
If you need access to cards installed in the machine, you can forget
running Asterisk under VMware. VMware
I don't know exactly what help you expect to receive in this forum.
Asterisk itself has nothing to do with VPNs of any kind, and you should
take your questions regarding the setup and configuration of them to the
appropriate place.
On 09/09/10 18:26, Deepika Nijhawan wrote:
I am not
On 09/12/10 07:06, Zeeshan Zakaria wrote:
I think this may be because ...
So you think, don't know. Maybe you knew if you knew the FreePBX
code, or bothered to look into it.
For God's sake, stick a sock in it. Others are attempting to help. You
are not.
--
On 12/02/11 04:02, Bryant Zimmerman wrote:
I am running 1.8.3 and my BLF lights have stopped working. The hints
appear to be intact when I use core show hints. But none of the phones
are getting the BLF updates. This has happend in the past and I have
had to restart my server. What could be
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