bristuff-0.2.0-RC8s
two isdn lines plugged into first two ports
and like I said, also a digium tdm400 card in there for analog phones
this 'timer' error message it is something to do with the qozap driver
isn't it? not sure
Thanks for any ideas!
--
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Tzafrir Cohen [EMAIL PROTECTED
] Error 2
and dumps me back to the prompt.
I am working with a fresh install of fc6.
Any help is appriciated.
Chris
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C:919-820-5473
On 3/16/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Chris Nighswonger wrote:
I am working with a fresh install of fc6.
Kernel 2.6.20 was released after Zaptel 1.4.0, so it will not build
against that kernel. Either use an older kernel, use the SVN version of
Zaptel branch-1.4, or wait
and as far as I know, no isdn
wiring has been changed or anything
ideas, appreciated!
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On 3/8/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Thanks for the responses.
iptables on the * box has no rules and all tables default to 'accept.'
I have not got to the point of placing calls out across the internet
yet. The issue here is no audio back from the * box when running
through
His vindictive dialer isn't playing while it is listening to rings or
busy signals.
Forgive my ignorance, but what on earth's a vindictive dialler? Is it one
with a strong sense of revenge? :-)
Regards,
Chris
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of 130ms+.
Are there any recommendations as to phones with particularly good buffering
that might iron out at least some of the poor network performance?
Thanks in advance.
Regards,
Chris
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/Polycom+auto-answer+config
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Sune Kloppenborg Jeppesen wrote:
They seem nice but with a Via EDEN 300Mhz CPU are they any more powerfule than
the Soekris net4801 with a 266Mhz CPU?
I get the version with a 1GHz CPU.
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not
allow setting that.
Is there any way to achieve this?
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based firewall/asterisk machine capable of running 20+
extension office. I have several installations like this.
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Brian Capouch wrote:
What kind of money are those things? There doesn't seem to be any
price information on the website you linked to.
About $350.
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Lacy Moore - Aspendora wrote:
Has anyone tried this?
This is how I do serious installations. I use a cheap ADSL connection
for browsing traffic and a dedicated feed for voice. Dual hjome the
machine, run shorewall setup for two ISPs, and write the rules to route
accordingly.
--
Chris
Is is possible to check voicemail by dialing one's own number?
When the outgoing voicemail message begins, I'd like to be able to
press some key and have it prompt to enter the password for that box.
Is this possible, and what option do I need to enable to make this function?
I searched google for asterisk voicemail documentation and could not
find anything.
After more searching, I found someone who had done it.
If you create an a extension in the current context, it will be
called when someone presses the asterisk during the outgoing message.
--
Chris Carey
On 3
.
Any thoughts? Or do I need to post more details?
Thanks,
Chris
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outside tomorrow.
Chris
On 3/8/07, Leonardo Kamache (Gmail) [EMAIL PROTECTED] wrote:
Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box.
Best Regards;
Leonardo Kamache
On 3/8/07, Dovid B [EMAIL PROTECTED] wrote:
If both the asterisk server and the softphone
I think you can do this with outlook. Use the Third Lane dialer product,
set your extension to that of the conference, then initiate the calls.
It will call the extension then the party and connect the two.
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se.rt.3.enabled=1
se.rt.3.ringer=11
se.rt.3.timeout=1000
se.rt.3.type=ring-answer
se.rt.3.name=Ring Answer
/
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I have used www.voiptalk.org for a number of years with their IAX2
connectivity and they seem very reliable with no echo issues. They will also
change the CID to your number if you fax them proof of ownership.
Chris
- Original Message -
From: --[ UxBoD ]-- [EMAIL PROTECTED
I'm still stuck on just exactly where in my extensions.conf file I
should put the code below. I'm running 1.2.14 of asterisk.
Chris Griffin
[EMAIL PROTECTED]
On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module. He gives the follow as an
example line
I'm still stuck on just exactly where in my extensions.conf file I
should put the code below.
Chris Griffin
[EMAIL PROTECTED]
On Feb 28, 2007, at 9:55 PM, Patrick wrote:
On Wed, 2007-02-28 at 23:28 -0600, voiplist wrote:
Thanks for the link..
As for Google, I know how to use it. I
it?
Thanks,
Chris Griffin
[EMAIL PROTECTED]
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There is nothing like that in my extensions.conf file. Maybe I should
have mentioned I'm running 1.2.14.
Chris Griffin
[EMAIL PROTECTED]
On Feb 28, 2007, at 3:07 PM, Mike Lynchfield wrote:
try putting near the exten = 1000,1,dial stuff
On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote
ahh! I am having this same problem all of a sudden
I've installed many TDM cards before ..never had this problem
what gives?
Trying to load zaptel 1.0.10 ...
Rev. G card ... tried uncommenting the revH fix in zconfig.h ...but no go
ideas?!
--
Chris
Michael C. Cambria [EMAIL PROTECTED
-recorded
files. Anyone have an idea what might be going on? The only problem is the
playback of .gsm files.
--
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[EMAIL PROTECTED]
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agent monitoring screen?
curious,
which app are you using for that?
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Chris Earle
Julian Lyndon-Smith [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Kyle Sexton wrote:
Started playing with 1.4 and I'm curious what uses people have come up
with for the Jabber integration? So
exten = s-CONGESTION,1,Congestion
[macro-uridial]
exten = s,1,NoOp(Outbound SIP URI call ${ARG1})
exten = s,2,SetCIDNum(0123456789)
exten = s,3,Dial(SIP/${ARG1})
exten = s,4,Congestion()
HTH
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IT Services | Fax: +44
with this would be a real life saver.
Thanks - Chris
From the zaptel-1.2.13 directory I issue the make linux26 command with the
following result:
make: *** No rule to make target `linux26'. Stop.
Just issuing the make command does seem to work and concludes with:
make[1
settings will be different for different users, but a
starting point that folks have found working well over low-cost ADSL
connections would be much appreciated.
Thanks in advance.
Regards,
Chris
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accept faxes in
on ISDN card and save on asterisk system ...? keeping digital signal strong
...
ideas appreciated!!
--
--
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System Solutions Specialist,
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in Germany, Sangoma in UK, etc)
If it's expandable through usual package addons etc, then it would seem
there is alot of added value because of the increased EASE of administration
over your well-maintained debian box
Thoughts?
--
Chris
Maxim Veksler [EMAIL PROTECTED] wrote in message
news
Wondering if you ever got this change made and if it did anything?
Update us if you please :-)
--
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Chris Earle (CBL) [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
thanks for your helpful investigation! I await news :-)
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- Original Message -
From
will find that the issue is variable jitter on the
line. The link above should help you figure this out
If this is the case, would upgrading to 1.4 with the new SIP jitter buffer
help at all?
Regards,
Chris
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, since they are
behind NAT. Each device re-registers with asterisk every 5 minutes to
prevent any possible NAT state timeouts.
Any pointers/places to look for potential problems would be much
appreciated.
Regards,
Chris
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(1:1, 5:1, etc.) is only available in a few exchanges in major cities.
Chris, if your customers are in the western US then please contact me
about dedicated circuits.
About 4500 miles away. :-) Thanks for the offer anyway though.
Any further thoughts would be gratefully appreciated, especially
,
Chris
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, the other had a Sangoma
A200. The A101 system has been rock solid. Could this be hardware related?
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+ handsets we have and scrap the MICS pbx and build an asterisk box with
our incoming POTS lines into a digium/sangoma card.
Any testimonials about these approaches? (Quintum boxes or Citel handset
gateways) Please share!
--
Chris Earle
System Solutions Specialist
Steve Totaro [EMAIL PROTECTED
Can anyone confirm that it actually works in Singapore with Busy Detect?
I have a system with loopstart and BusyDetect and have recently attempted to
improve disconnect detection results with the addition of
hanguponpolarityswitch ... results are mixed
--
Chris Earle
System Solutions Specialist
Linux on
2007-01-13 18:31:56 UTC
Asterisk Queue Logger restarted
Rotated Logs Per SIGXFSZ (Exceeded file size limit)
Feb 5 08:43:00 WARNING[20103]: format_wav.c:247 update_header: Unable
to find our position
== Parsing '/etc/asterisk/logger.conf': Found
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at that time?
Chris
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; option astlogdir
;
[logfiles]
;
; Format is filename and then levels of debugging to be included:
;debug
;notice
;warning
;error
;verbose
;dtmf
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to erase
them. I believe this is a bug but I don't see many other people report
it. I have seen a couple of instances of it, though.
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Bruno Castelo Branco wrote:
Hi
Try that one http://www.counterpath.com/index.php?menu=Productssmenu=xlite
Bruno C. Branco
Do they still have the web-based one available, formerly X-Web?
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Your dinner's in the oven.
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...to work in the UK
If removing helps the disconnect detection, I don't mind losing my callerID
support.
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Ed W [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Chris Earle (CBL) wrote:
Sorry -- you're right, I didn't express the scenario properly ..
The disconnect
-8
-=-=-=-=-=-=-
-=-=-= Zaptel.conf =-=-=-
fxsks=1-2,5-6
fxoks=3-4,7-8
loadzone = uk
defaultzone=uk
Ideas?
--
Chris
Carlos Rojas [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hello,
What's your zapata.conf and zaptel.conf?
On 1/20/07, Matt Brown [EMAIL PROTECTED] wrote
this
or anything similar.
-Chris
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thanks for your helpful investigation! I await news :-)
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- Original Message -
From: Matt Brown [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Saturday, January 20, 2007 7
of premature hangups
Thanks for your query
--
Chris
- Original Message -
From: Ed W [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, January 19, 2007 1:26 PM
Subject: Re
I can honestly say Unwired Buyer's competitive advantage is definitely not
the fact that they use Asterisk or Allison. Those two things were/are
definitely development advantages though.
On 1/22/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Jens Vagelpohl wrote:
There was a link posted to an
Unwired Buyer paid for ExternalIVR. At this point, they're not owned by
eBay.
On 1/22/07, Colin Anderson [EMAIL PROTECTED] wrote:
They do. The ExternalIVR application was developed in cooperation with
them.
lol so the $2.6b they spent on Skype was well worth it, then. It'd be
nice,
though to
for CPC.
Does anyone have any thoughts/confirmation about this finally being a viable
solution? This disconnect supervision problem has plagued TDM and Sangoma
cards for a long time!
Comments appreciated before I get on the phone with BT
--
Chris Earle
System Solutions Specialist
is intended to ALLOW
caller input (as in an IVR) while sound is playing I guess I am
confused as to the Background command's purpose ...
Any ideas for what I want to do ?
--
Chris
[EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Thanks for your reply,
But I want to have
the Elmeg and the
target destination.
Can anyone shed any light on this?
Thanks in advance.
Regards,
Chris
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on this?
If it is available, is there any documentation on the feature?
-Chris
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the change of Telus' (the
ILEC) customer service system entirely to speech recognition. It
actually works really, really well I've never been able to screw it
up
What happens if you yell I just want to talk to a human being! really
loudly at it? ;-)
Regards,
Chris
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suggestions? I know the hardware works, because it was working as my
asterisk 1.2 test system before I reloaded it completely and installed
asterisk 1.4
I appreciate the help.
-Chris
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app_system.c:105 system_exec_helper: Unable to execute '/sbin/zapscan.bin'
when I log into the web interface.
Any ideas?
-Chris
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Ok. I finally got past this. After doing all the relevant udev stuff, I ran
a make config from the zaptel sources, and got the service to install.
I'm still quiet an asterisk newbie, and defiantly a huge Linux newbie, so
thanks for the help.
-Chris
Eric ManxPower Wieling wrote:
Chris Miller wrote:
I would tend to agree, but the context that holds these number is an
inbound context which includes additional logic that would fail
normal calls. Yes, I can add the DIDs to the outbound context, but
the point here is not to have a bloated
I've read), but I cannot find
out how to resolve this. I've reinstalled zaptel several times. I read a lot
about having to read the README.udev file in the zaptel source, but I don't
even have that file on my system.
If anyone has any ideas I'd love to hear from them.
-Chris
there ought to be an easy way get Asterisk to consult it's
own inbound DID routes before selecting an outbound trunk, and without
populating the dialplan with a parallel list of DIDs. I can't imagine
I'm the only one to have run into this, but there's nothing on the lists
about this scenario.
Chris
don't feel
comfortable trying to hack the patch code into the current version.
Just trying to avoid reinventing the wheel if there's already a known
workaround.
Chris
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Thanks for the input
not seeing any evidence of disconnect supervision or callerid but
you're right, depends on provider ---
Anyone have any luck with provider MTNL?
--
Chris
Rajkumar S [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
On 1/1/07, ram [EMAIL PROTECTED] wrote
I would like to convert a file to WAV49 for use with Asterisk using
the linux command line. Specifically I would like to upload sounds to
use for unavail.wav and busy.wav, but I'd like them to be compressed
so that space is not wasted.
I tried using SOX but havent found the correct command-line
Greetings list,
Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even with
the ~amd64 keyword, latest in the official Portage repository is 1.2.13.
Thanks in advance.
Regards,
Chris
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Hey all,
anyone know the status of disconnect supervision on POTS lines in India?
Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have
disconnect supervision..
Thanks
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System Solutions Specialist
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Hi List
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Thanks
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Entropy IT Ltd
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PROTECTED] *On Behalf Of *Chris Johnson
*Sent:* Monday, December 18, 2006 6:07 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] BLF on GXP2000
Well, I am making some progress. I have made some changes as defined below
and now have a green line
On 12/18/06, Eric Jacksch [EMAIL PROTECTED] wrote:
Is the problem just when you don't answer the cell phone? Many cell
phones go to a voice announcement when they're turned off or not answered,
and Asterisk thinks the call has been answered. The other issue could be
that your gateway
coffee.
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Chris Johnson
*Sent:* Sunday, December 17, 2006 4:50 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] BLF on GXP2000
I am trying to set up the BLF on a GXP2000
The problem I am running into is that when the call to my cellphone is made,
it appears as though the call completes so it never rolls to asterisk
voicemail.
Here is my current config:
exten = 102,1,Dial(${sipura},10,)
exten = 102,n,playback(pls-wait-connect-call)
exten =
I reload the phone, I get the following error:
[Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.248, but there is
no
hint for that extension
Any help is greatly appreciated.
Chris
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten = 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED] )
exten = s,102,VoiceMail([EMAIL
that smokeping gives a
pretty good representation of how call quality will be.
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I have correct Caller-ID information coming in on the 'Remote-Party-ID' header.
The From value is being sent in as Unknown.
How could I replace the From value , or CALLERID(all) with the correct
values that are in Remote-Party-ID? Or is there a way to tell asterisk
to read that header?
I like smokeping, as it gives a good sense of the quality of the route
over time.
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Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
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In this case, the machine was a spandsp virgin, it had never been
installed before.
I made sure I ran ldconfig before and after building, and still no joy.
I have managed to get iaxmodem and hylafax to work quite well though :-)
Chris
On Fri, 2006-12-08 at 12:43 +, Steve Davies wrote
me where I'm going wrong, been trying to get this
to work for hours. I've got rid of all the old libraries, recompiled...
my next step is to sacrifice a goat!
Any help greatly appreciated.
Chris
--
Chris
--
E Mail: [EMAIL PROTECTED]
SIP: [EMAIL PROTECTED]
IAXTEL
this approach, but my agents are not being
logged off automatically using autologoff=20.
Any help to easy my lack of sanity would be greatly appreciated
Best regards,
Chris
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, but this is failing as the
variable appears to be lost when the call goes back to the queue.
Can anyone suggest an answer to this puzzle for me.
Many thanks
Chris
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, but that
was with SIP registrations, not IAX. A firmware update fixed that.
I'm no * dev, but what exactly is so crap about IAX?
-Chris
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, and extensions.conf.
I would be most grateful if someone could show me the way.
Thanks in advance:
Chris
Asterisk ver: 1.2.10
PHP:
#!/usr/local/php/bin/php -q
?php
$stdin = fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/var/log
Forum wrote:
Does anyone on this list know of a reputable T1/PRI provider in St.
Lucia? If so, what monthly costs am I looking at? I do know that
Cable and Wireless are the biggest Telco.
I think you will find they are the only telco and the cost will be enormous.
--
Chris Mason
(264
can see asterisk playing the sounds but
my phone goes from ringing to busy, and I don't hear the phontics.
Below are the relevant bit from my PHP, Console, and extensions.conf.
I would be most grateful if someone could show me the way.
Thanks in advance:
Chris
Asterisk ver
call that queue, the
phones already on calls return SIP BUSY,whilst the others ring as normal.
It's not perfect, but for most of our users the call waiting noise in the
earpiece is an annoyance anyway.
Hope that helps.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email
accounts, and nobody has gotten back to us about
this... FOR THREE WEEKS! Is there anyone else out there that has dealt
with them before? If so, I'd like to hear your opinion.
Thanks,
Chris
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asterisk
Andrew Joakimsen wrote:
Chris:
We were evaluating AgileVoice currently, could you please elaborte on
your problems? Did they not do the instattion for you?
They did the installation.
I'm going to be very careful with my wording here, but if you are
currently evaluating their software, I
-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Chris Mazuc
Systems Administrator
DataGroup Technologies
(252)329-1382
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My company has had a few screens go out on us, but all of those were
completely blank. I'm not sure if we just got a bad batch or what, but
the Snom phones are usually a solid piece of hardware. I'd try to RMA it.
Nick Hoffman wrote:
Hi guys. I just bought and configured a Snom 360 and have
This was orignally posted on The Asterisk Blog Forums. See the original post here.Pete101 says: I am having issues with all inbound calls coming
into the system. It is taking like 10 seconds for it to decide where to
route the call. It applies for both PSTN calls and VoiP calls. Does
anyone have
to see if there is a way to blindly accept calls from a known
IP address, but I don't think there is a way that would retain CDR
information.
Chris Mazuc wrote:
I have an asterisk box at a remote location (which I will call remote),
which registers to my local asterisk box (I'll call that one
it seems to
be ignoring the username part... or maybe I need to go read some RFCs.
Any help is greatly appreciated.
Thanks,
Chris Mazuc
-- SIP read from REMOTE:1025:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;rport
From: My Name sip:[EMAIL PROTECTED
, and it should work for you. Both outbound routes can use the same trunk without issue.
AlexOn 10/17/06, Chris Ramsey
[EMAIL PROTECTED] wrote:
This was posted at The Asterisk Blog Forums
Click here for the original post.
I need someone to explain how the dialplan rules
work? I'm having a hard time
This was posted at The Asterisk Blog ForumsClick here for the original post.
I need someone to explain how the dialplan rules
work? I'm having a hard time getting it. I know that to dial out you
need a 9 and to ignore that 9 once your out... requires a switch of
sorts that tells asterisk to ignore
I agree with Mohamed. TrixBox is an excellent way to start, but in the long run, if you attempt to use Asterisk in a business setting, you will probably want to be able to hardcode the conf files yourself. I have only recently changed over to TrixBox from a standard installation on a debian
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