the others work fine, it's just this one that doesn't.
The box in question is running 1.4.22, but I have had a similar issue in the
past with a 1.2 box, so it does not appear to be version specific.
Any thoughts?
TIA.
Regards,
Chris
A really, really quick question here!
Am I right in thinking that all passwords sent across the network in
Asterisk are MD5 encrypted without me having to specifically set anything up
to make it happen?
Thanks so much!
Chris
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A really, really quick question here!
Am I right in thinking that all passwords sent across the network in
Asterisk are MD5 encrypted without me having to specifically set anything up
to make it happen?
Thanks so much!
Chris
Sorry, I forgot to add when using the SIP protocol
On Mon, Feb 9, 2009 at 9:28 PM, Kevin P. Fleming kpflem...@digium.comwrote:
Chris Rowson wrote:
Am I right in thinking that all passwords sent across the network in
Asterisk are MD5 encrypted without me having to specifically set
anything up to make it happen?
The simple answer is 'yes
provided. I thank you
for your time, and effort, and I look forward to hearing from you.
Regards,
Chris.
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I can find FANLESS 24 port PoE 10/100
That's an achievement in itself. Can you post details - I have quite a few
locations where that might be useful...
TIA.
Regards,
Chris
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for an ATA too ;-)
Chris
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To be fair, this is likely a little out of scope for an Asterisk discussion
list, but you might get more help over at the Centos website
http://www.centos.org/
Have fun!
Chris
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*1) What name I have to save it.Like what extension ?*
extension? that isnt important but the common is to use .sh for shell
scripts
That's one of the strange things you'll notice if you're used to a Windows
environment. Under Linux, it doesn't matter if you give your script a file
You are configuring Asterisk to LISTEN on 5062 , if you want it to talk
to another server on 5062, then configure that server's config stanza
accordingly.
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My dial plan includes a number of service codes that include #, largely
because I wanted to mirror BT's codes as I know them already. Example,
check the divert on an an extension: *#21 (as opposed to BT's *#21#).
Everything was working fine.
I've just got around to setting up DISA and find that I
},provider1,provider2)).
Thanks to all who replied. Looks like I just need to do a bit of
extensions.conf find/replace then.
Any thoughts on the CDR issue?
TIA.
Regards,
Chris
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for pgsql from voip-info.org, which, again, has worked fine
logging 1.2 and 1.4. Have there been any schema changes in 1.6 one needs to be
aware of?
Thanks in advance!
Regards,
Chris
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to mirror the full VoIP
VLAN to the Oreka port wouldn't I? Or is there another reason that
I'm missing here?
Chris,
Make sure that all of your SIP clients are set to canreinvite=no in
sip.conf. The default is canreinvite=yes, which allows RTP to
bypass Asterisk. Certain things
to the Oreka port wouldn't I? Or is there another reason
that I'm missing here?
Just trying to get this sussed out in my head!
Thanks for your time.
Chris
Hi again, didn't get a reply to this one. I'm a bit stumped so I thought I'd
try the list one more time to see if anyone has an answer
wouldn't I? Or is there another reason
that I'm missing here?
Just trying to get this sussed out in my head!
Thanks for your time.
Chris
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the problem nicely.
If that's not an option, how about writing a transfer macro that'll return the
call to the originating extension if the transfer is unanswered within X
seconds?
Regards,
Chris
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?
Any suggestions?
Chris
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If you don't like the forward key why not simply get rid of it.
With firmware 3.1.0 all you need to do is add one line to your config file:
softkey.feature.forward=0.
While you are at it, you might (or might not) like to get rid of buddies and
mystatus.
softkey.feature.buddies=0
to set it up today as my
experience is so out of date, but I can offer you encouragement that it
will work for you if you put in the effort. There is nothing fundamental
which will cause it to fail. Which carrier are you planning on
connecting to?
Chris
Moises Silva wrote:
Hello Peter,
You
make these discussions go easier.
Chris
Peter Lindquist wrote:
We are using TOT (Telephone Organization of Thailand). They are very
messy on their side so we are sorting out some unrelated problems with
them right now - very slow response.
I believe you are correct when you say
/or wherever) and then
loads the completed file into that directory, or if it writes the file
to the directory directly, appending it till the recording is
finished?
Hope that makes sense!
Cheers
Chris
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if this is where the file is created, and then moved to the INBOX
folder perhaps?
Chris
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directory (since .txt files are
updated after the message is complete).
--
Tilghman
Thanks Tilghman, that's really helpful :-)
Chris
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-chance
that someone might happen to know the answer off the top of their head
so I could get back to someone else! I'll just try it later when I do
have access.
Chris
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asterisk
recording files, write them first into a temporary holding
directory, then move them into the final location when done.
I'll give it a try and see what it does.
Chris
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Ubuntu and get stuck with anything, you can use
the Ubuntu 'Answers' helpdesk where a community member will try and
guide you through any problems you may have.
https://answers.launchpad.net/ubuntu/
Cheers
Chris
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are others too...
Chris
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in 1.2 ?
Thanks in advance.
Regards,
Chris
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be
pretty similar between the models.
Regards,
Chris
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a pretty web interface, too). The pretty web interface is less fussy than
the Netgear one (which seems unreliable in non-Internet Exploder browsers).
On the other hand, the Netgear is substantially less deep (an issue in some
wallmount cabinets) and definitely a lot quieter.
Regards,
Chris
.
Regards,
Chris
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and how it works, I'd help out, but alas, I
don't, and don't really have time to learn.
For some of the packages you need to edit the ebuild file and add ~amd64 ~x86
(depending on your architecture) to the ARCH= line, then rebuild the digests
on the ebuilds.
Regards,
Chris
to register via
the website first, entering a payment card to be used for future orders, then
give then a customer number and PIN that can be used by telephone for future
top-up orders?
Something like that would be fairly easy to query against a database. I'd have
thought.
Regards,
Chris
the issue entirely.
The calls in question are being delivered to the boxes via IAX or SIP, so it's
not a Zaptel gain issue.
Has anyone else had similar problems? Is it safe to drop the silence
detection threshold even lower?
Thanks in advance.
Regards,
Chris
You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in
the overlay.
# emerge layman
# layman -a voip
You may need to modify some of the .ebuild files, or your
/etc/portage/packages.unmask depending on your asterisk build.
Regards,
Chris
that also
offers similar training without having to be present at the Bootcamp
That way someone could elect to train at their own schedule and later
coordinate to drop-in on the last day of a Bootcamp session and take
the dCAP.
- Chris
On Sep 21, 2008, at 9:15 AM, Tilghman Lesher wrote
, at 10:20 AM, Chris Brentano wrote:
I would also disagree that the written exam is biased towards people
who attended the training. I attended a Bootcamp earlier this year and
thought I was fully prepared to pass the dCAP. Especially since I
already had real-world Asterisk experience
+features.conf
Regards,
Chris
2008/9/18 Gergo Csibra [EMAIL PROTECTED]:
Hello asterisk-users,
My SIP phones are in pickupgroup, and if some of them ringing from
other phone can pick up with *8 as usual. But I want to know if this
happen. I've tried the a extension, but seems not working.
Any
in advance.
Regards,
Chris
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on laptops.
Not directly on-topic for this list, but I'd not heard of OpenSIPS before, so I
had a look at the website. It looks to be a fork of OpenSER. Does that mean
OpenSER development has slowed/ceased, or has the OpenSER project itself
morphed into OpenSIPS?
Regards,
Chris
I don't really see the advantage to be honest. If I needed Asterisk access from anywhere I'd just run it locally on my laptop, connect to an ITSP via IAX or SIP, and run a softphone app locally. The only exception I can think of is when you'd want people to be able to leave voicemail on your
,
Chris
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I tried DTMFmode=auto and it did not help. Any further ideas?
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Hi,
You can find some info about differences between 1.4 and 1.6 here:
http://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE.txt?view=markup
Kind regards,
Chris
2008/8/28 --[ UxBoD ]-- [EMAIL PROTECTED]:
Hi,
I would like to give 1.6 a try and was wondering about the configuration
:
dtmfmode=rfc2833
and in the [general] section: relaxdtmf=yes
I have a very similar system at my office nad DTMF works.
Any ideas why this does not work?
Chris Mason
Comet Systems
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Ruchir wrote:
Have you set dtmf mode rfc2833 or avt in your phone?
No, I have not changed anything in the phone. The sip.cfg setting is the
default:
DTMF tone.dtmf.level=-15 tone.dtmf.onTime=50
tone.dtmf.offTime=50 tone.dtmf.chassis.masking=0
tone.dtmf.stim.pac.offHookOnly=0
Jonathan Disher wrote:
He has two buildings (the office,
and the shop proper), separated by about 3-400 yards.
Your inter-building distance exceeds ethernet over copper limits, you
will need a fiber link.
paging intercom (to
page employees, etc) on a dedicated extension -
Easy to
system. If the same problem occurs,
then you can narrow it down to the existing card or system. Then just
narrow it down further. I'd probably then try the existing card in the
new system, just to verify whether the card is damaged.
Good luck.
-Chris
On 25 Aug, 2008, at 7:27 AM, Jakub Arkon
a different
SIP local port for each account, but the Zoom unit does not appear to
allow the SIP local port to be specified on an account by account basis.
Can I get the unit to register two separate accounts on Asterisk from
the same port and IP?
--
Chris Hastie
Off the top of my head... you could probably route the audio of a
softphone (like Zoiper/X-Lite) to something like Nicecast (Mac) or
Icecast.
On 11 Aug, 2008, at 9:21 AM, Andrew Niemantsverdriet wrote:
Hi all,
I have an interesting problem that I am looking for a solution for. I
want to
I am using Asterisk 1.2.24. I know it should be upgraded, but that is
not an option at this point for this working system.
I am experimenting with using an external application to control whether
a call should be connected. Most of the time it works. Sometimes the
dial plan never comes back
I'll answer my earlier question regarding System commands and zombie
ringing channels that can't be killed.
If I start with Answer in the dial plan, I don't have the problem.
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I have two Asterisk 1.4 boxes connected via IAX over a VPN tunnel on a
10Mbit link. We never did any stress testing as it's a temporary
arrangement, but we've never had any call quality issues or run up
against concurrent call limitations. I'm mostly routing internal
extensions over the
someone else down the line.
Me too,
Any reason you want this off the list particularly?
Chris
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I would look at AIPHONE for product that are built to do this and work
perfectly.
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not always detected, DTMF not always detected, etc.), so
it's probably time to look at fully IP alternatives.
Any suggestions gratefully appreciated.
Regards,
Chris
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Greetings list,
Is ARI by Littlejohn Consulting still being maintained? Their website seems to
have disappeared, and the only download links I can find through googling are
back in 2005.
If it has disappeared, what are people using as alternatives these days?
Regards,
Chris
think of right now. Can all these issues be dealt with by:
1 -- a sort of easy route, add a CAPTCHA to the web form
2 -- compare against lists, or somehow do asterisk dialplan logic to stop
wellhow could you stop legit numbers? :-S
Ideas, suggestions appreciated!!
--
--
Chris Earle
should be coming from and it now works.
Working file ---
[from-pots]
exten = 277,1,Answer()
exten = 277,n,Wait(3)
exten = 277,n,Playback(tt-weasels)
exten = 277,n,Hangup()
So in summary it was basically me misconfiguring the box...
Cheers
Chris
asterisk console via 'asterisk -rv'
and see nothing).
I wondered if it might be a problem with Asterisk not listening
properly, or perhaps a problem with my home firewall. Would anyone be
kind enough to advise me as to where I may have gone wrong?
Thanks, Chris.
My sip.conf looks like
with Asterisk, you should configure it as
an H.323 gateway. Why are you trying to set softswitch?
That is how all of our systems are configured with Asterisk and ooh323.
Works very well and very stable.
Chris
Everton Goularth wrote:
Hi people,
Someone have already used asterisk with Nextone?
I`m
: [EMAIL PROTECTED]
Content-Type: application/sdp
CSeq: 1 INVITE
From: Chris Nsip:[EMAIL PROTECTED];tag=5250369537080
Max-Forwards: 70
To: sip:[EMAIL PROTECTED]
User-Agent: SJphone/1.60.289a (SJ Labs)
v=0
o=- 3422571157 3422571157 IN IP4 client.internal.ip
s=SJphone
c=IN IP4 client.internal.ip
t=0 0
have used many fsm7326p to power 24 phones or 726tp to power 12
phones and they work great
On the Linksys side, we have a load of SRW-224P switches out in the wild
powering 24 Snom 370s (around 7W each) off each switch.
Regards,
Chris
or some
abysmal pricing on the 300s :-)
Regards,
Chris
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be very grateful.
Chris
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I don't really know as I am unfamiliar with BRIstuff. If fact the whole ISDN
world is a new one for me as its not that common for small business here in
Australia. We are using the Digium B410P. A quick Google of B410P and
BRIstuff is inconclusive.
-Original Message-
From: [EMAIL
of how could one determine the actual ring voltage, as well
as any other analog line values that would help troubleshoot this
sort of issue? If there are further issues with the analog lines, we
may need the ability to detect and tweak the ring detection parameters.
Thoughts?
Chris
That what I wanted. Extension 0 in the dialplan connects to the
console. The Async option lets me call an extension from the console
using a manager API app. The only downside is that by reversing the
originate, the console doesn't hear any ring back. The 100% CPU
utilization in the first
When I initiate a call from the console (Console/dsp) to a local SIP
extension, asterisk uses up 100% of the CPU until the extension
answers. It happens when using .call files or the manager API. My
examples are for the manager API, but .call files perform the same way.
Here is the 100% CPU
Generally I'd agree. But it could at least more adequately notify the
user, even if they are compiling on a different system than where it
will be running on. It just seems that in most cases people will be
compiling on the system they will be installing on. This is what they
teach at the
PKG_CONFIG_PATH=/usr/lib/pkgconfig and reran
./configure and it was happy again.
- Chris
Tzafrir Cohen wrote:
On Sat, Apr 19, 2008 at 11:11:53AM -0700, Chris Brentano wrote:
Nevermind, I found the problem.
And for the benefit of the readers of the archives: what
I believe this isn't a Polycom thing, but the nature of SIP devices in
general. But, that said, Polycom should start making IAX desk phones. :-)
- Chris
Lee, John (Sydney) wrote:
DND does not do anything for me BLF-wise either (shame). Simply
picking up
the handset won't do
BTW, I did this and it did not work unfortunately.
My /etc/ld.so.conf looks like:
include ld.so.conf.d/*.conf
/lib
/usr/lib
shrug
- Chris
Tzafrir Cohen wrote:
On Sun, Apr 20, 2008 at 09:07:09AM -0500, Tilghman Lesher wrote:
Tzafrir Cohen schrieb:
On Sat, Apr 19, 2008 at 11:11
Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start up
Asterisk (with -cvvv) I get an error regarding func_curl.so
(lines omitted)
...
== Registered custom function STRFTIME
== Registered custom function STRPTIME
== Registered custom function EVAL
== Registered custom
Nevermind, I found the problem.
Chris Brentano wrote:
Asterisk 1.4.19, Zaptel 1.4.10 and libpri 1.4.3. When I try to start
up Asterisk (with -cvvv) I get an error regarding func_curl.so
(lines omitted)
...
== Registered custom function STRFTIME
== Registered custom function STRPTIME
Mike wrote:
do a decent
job of providing QoS on the upstream, which is where you (usually) need it
anyways.
QOS can only be on outgoing, you can't set the priority of a packet
after you receive it. The only other solution would be the cooperation
of the ISP to provide QOS upstream of you.
/suggestions gratefully appreciated.
Regards,
Chris
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Additionally Mark, a Channelized (also called Integrated) T1 offers 24
channels for voice/data, but after bit robbing (for signalling, etc) you
only get around 56kbps per channel. ISDN PRI over T1 has 23 b-channels
of voice/data and one d-channel for signalling, etc. PRI is preferred
and most
at this stage - they're mainly to take registration load
away from Asterisk.
Regards,
Chris
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.
trixbox.org (rpm based)
Bit too heavy for what I'm after.
Thanks!
Regards,
Chris
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Greetings list,
Not exclusively asterisk-related, but I've noticed the CentPBX site has been
offline the last few days. Anyone know the reasoning behind that, and more
importantly, is anyone mirroring it?
Thanks in advance.
Regards,
Chris
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after for this deployment.
TIA.
Regards,
Chris
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at the start of the call right? once that occurs, the EC is
kicked in and everything is fine?
--
Chris Earle
System Solutions Specialist,
Network Technologies Division
CBL Data Recovery
w: http://www.cbltech.com
Rob Schall [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
We have a location
any
asterisk code.
--
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On Tue, Mar 25, 2008 at 9:54 AM, Chris Carey [EMAIL PROTECTED] wrote:
Has anyone put together a patch which would allow a different
fromstring and emailbody based on context? Or any other way to have
more than one fromstring and emailbody per server
Has anyone put together a patch which would allow a different
fromstring and emailbody based on context? Or any other way to have
more than one fromstring and emailbody per server?
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Phone Book, etc. etc. all without any issue.
Regards,
Chris
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Markham Facility
http://www.cbltech.ca
Be committed to the environment. Please think twice before printing this
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What do you recommend I do so both AGI scripts and PHP scripts can
work with a common SQLite file? Should I run Asterisk as www:www,
www:wheel? Something else?
I run the web server and apache both as the user asterisk
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What do you recommend I do so both AGI scripts and PHP scripts can
work with a common SQLite file? Should I run Asterisk as www:www,
www:wheel? Something else?
Correction: I run the web server and asterisk both as the user asterisk
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handsets
seems to lead to all sorts of stability issues
Regards,
Chris
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find it a bit quiet on the sound quality, sometimes a problem with
background noise.
Should be an option in the web interface to adjust volume - I set all the ones
I deploy to high
Regards,
Chris
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only 1
line:
exten = [number]XXX,1,Dial(IAX2/someserver/${EXTEN})
Regards,
Chris
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of
the A - C queries.
Regards,
Chris
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/ext ... it won't pass that
variable along so all calls to external channels are considered Queue
calls.
:-\
gotta maintain that 'information' when I dial out over IAX2 to the other
peer
any random ideas appreciated,
--
Chris
Chris Earle [EMAIL PROTECTED] wrote in message
news:[EMAIL
They get the time from their NTP server
On Mon, Mar 10, 2008 at 11:59 AM, Don Smith [EMAIL PROTECTED] wrote:
I am running Asterisk 1.4.5 on a debian Linux server. Saturday night/Sunday
Morning Daylight Savings time occurred. The server shows Mon Mar 10
10:59:42 PDT 2008 when I do a date
, deregulated, competition, one
incumbent telco, etc.).
Regards,
Chris
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of the Swiss Telco market is non-existent.
Are there any folks on the list who've experience in this market who might be
able to offer some hints, or companies to approach?
Thanks in advance.
Regards,
Chris
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For full contact details visit http
and change voicemail
settings all from your iPhone.
The technology behind it is Asterisk (The Open-Source VoIP PBX), PHP
for the backend, Smarty and iUI for the frontend.
--
Chris Carey
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.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons
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