Well, this is mostly just new. I thought this would be a good place to announce the opening of the forums on The Asterisk Blog. Check it out and leave a few messages to help get it's feet wet. Thanks guys!
http://www.AsteriskBlog.com/forum-- www.AsteriskBlog.comYour home for easy to learn Asterisk
So I'm sure many of you are using or have tried to use TrixBox. Thus far, I'm in love with it. I haven't had a single snag. Then again, I don't need to get into anything overly nitty gritty with my Asterisk box.What are your views?
-- Want a free copy of TrixBox Made Easy?Read the contest rules
Hmm, I haven't seen that happen yet. Like I said though, I don't need anything really large and in charge. I have a very basic dialplan with a couple of Background's and VoiceMail apps. That's about it really.Do you have an example of something that it had written over?
-- Want a free copy of
If you have a site and don't have it on StumbleUpon, you may want to hitch a ride. I found the group when I found that Matt had put my site on the list and I was getting traffic from the website. It's not a HUGE amount of traffic, but any good traffic helps.
On 10/8/06, Matt Riddell (IT) [EMAIL
Hey guys, just thought I'd let you know that I'm giving away a copy of TrixBox Made Easy on The Asterisk Blog. Check it out.--
www.AsteriskBlog.comYour home for easy to learn Asterisk stuff.
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To
I don't think you would need a macro for this. After Asterisk determines that their first extention is unavailable, give the user the choice of trying the cell or just going to VM via a background command.
On 10/10/06, Dovid B [EMAIL PROTECTED] wrote:
You can create a macro that tells the caller
unreachable or goes offline ...then what happens to the CDR data for
that time period?
Suggestions appreciated
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own CID even then they
normally ask for proof of the numbers you wish to use.
Chris
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Hi,
Someone has a Grandstream GXV 3000 to run a small test with
me? I have one GXV 3000 setup and I cant get video from that videophone
with Eyebeam.
Best regards,
Chris HARIGA
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at bugs.digium.com appear to be unrelated.
Anyone seen this issue and know what is causing it?
Chris
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voip to pstn call), unless
frames were being dropped.
Chris
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Does Asterisk 1.4 support one server SUBSCRIBE for multiple voicemail
boxes? The individual phones are registered to a seperate server B.
That server B will send a SUBSCRIBE to Asterisk server A and server B
will handle passing the MWI information to the individual phones.
and know what is causing it?
Chris
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Has anyone implemented Asterisk voicemail to work with a Sonus system?
The Sonus is sending what they call an implicit subscribe where one
subscription should collect MWI information for all mailboxes.
Chris Carey
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locations
(since the DC also charges for that), but I'm not sure that's realistic
given the hardware we're looking to use.
Any thoughts gratefully appreciated.
Regards,
Chris
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Before the phone starts it boot process, I edit the phone settings under server
information, enter the IP of the sip server and also the protocol to use, UDP
Only.
Chris
David Gagnon wrote:
Hi Chris,
I'm would like to get more information about this problem. Why the
phone
Not to mention the feature that the new firmware and bootrom that prevent it
from registering with the Asterisk server unless you hard code the sip settings.
Chris
Jessee J Holmes wrote:
Also keep in mind that as of right now, the latest bootrom and firmware
available from Polycom
just figured out that I need to hard code the sip
server and tell it to talk udp only. After this, the phones worked again.
Any idea on what I need to configure to fix the phones so they will know which
server to talk to and only talk to it via udp?
Chris
?
Any suggestions appreciated
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have a space in the sip peer name i.e 'Chris Teesdale' Would have to be 'ChrisTeesdale' for it too work.
Regards
Chris Teesdale
I.T. Development
Philips
Tel : 01325 384394 ex 221
Email : [EMAIL PROTECTED]
On Wed, 2006-08-23 at 10:03 +0100, Simon Woodhead wrote:
We've done this with OpenVPN
. The
idea of rewriting every application in the system because EXTEN can't be
changed is very unappealing. I'm already 4 days late getting this up and
running.
Thank you for any assistance,
Chris
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understand is why it
would work outbound, but not inbound...
Any thoughts / comments / suggestions on this would be very helpful, and
I thank you in advance.
Cheers,
Chris
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Jeremy McNamara wrote:
as most people here know, I yell at stupid people.
Be honest, Jeremy, you yell at everyone!
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For HFC cards in the UK, check out Solwise - www.solwise.co.uk
Regards,
Chris
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any difference, I am using a grandstream 386 ATA
Thanks
Chris
On 8/6/06, Hadley Rich [EMAIL PROTECTED] wrote:
On Monday 07 August 2006 06:36, Chris Hembrow wrote:
I am new to asterisk, and learning as I plod along. Currently, I am
trying to work out how to create a ring group without using AMP
is shockingly bad, there should
at least be a warning about that).
All in all, not a good start :)
any help would be MUCH appreciated
thanks
Chris
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= _003764.,1,Congestion
Exten = _003765.,1,Congestion
This is just so long winded, and you can imagine doing this
for a huge list of destinations.
If any one can suggest an improved or more efficient way of
doing this, I would be greatly appreciated!
Best regards
Chris
--
Chris Blunt
.
Regards,
Chris
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through another application (or
through AGI scripts), but Asterisk is currently my only means of
communicating with the Junghanns.
Any suggestions?
Thanks!
-Chris
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Hi,
We purchase the database with zip codes, latitude, longitude, are codes and
all for our zip lookup AGI.
If you need something simple take a look at
http://www.census.gov/geo/www/gazetteer/places2k.html
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto
with a different IP address pointing to a different default gateway
(router). But then some how load balanced into a virtual network connection?
Any ideas or solutions would be appreciated just in
case I have gone off at a wild tangent.
Thanks
--
Chris Blunt
Entropy IT Ltd
shortest route.
What codec are you using with sellvoip? I have to use G729 but I find
that while the calls are setup, I get one-way audio on every call. The
called party cannot hear me. Let me know your config if you would.
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... error while writing audio data: : Broken pipe
Any ideas? I'm running Asterisk v1.2.9.1, on Centos with a Linux 2.6.15
kernel. There's an X100P in the system, so I'd assume timing isn't an
issue.
Rgds,
Chris Jones // Network Administrator
Top Level Internet
e: [EMAIL PROTECTED
, but it sounds like they may be
related to the Trixbox compile of the latest Asterisk.
Regards,
Chris
Chris Miller
President - Rocket Scientist
ScratchSpace Inc.
(831) 621-7928
http://www.scratchspace.com
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by using CHanIsAvail like this:
exten = s,n,ChanIsAvail(IAX2/teliaxIAX2/nufoneIAX2/sellvoip)
This way, I choose the historically highest quality provider first but
roll over if they are down.
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, you can log into Kall8 and change the forwarding number.
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I've just found that picking up another phones call via *8# gives me the call
but the other phone keeps ringing. Anyone else seeing this on svn head (updated
last Sunday).
Chris
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asterisk
()
exten = 0870xxx,3,Playback(cust-greeting)
exten = 0870xxx,4,SetCIDName(Tech)
exten = 0870xxx,5,Dial(SIP/4902,15,tr)
exten = 0870xxx,6,Dial(SIP/4902SIP/4903,15,tr)
exten = 0870xxx,7,Voicemail(u7003)
exten = 0870xxx,8,Hangup
Thanks for your time and advice.
--
Chris Blunt
to configure for Asterisk.
Also, set the dftm mode to be commented out, then the tone generation works.
;dtmfmode=rfc2833
Also, I beleive CallerID does not work properly on some firmware versions.
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Chris Mason
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Marcin Lukasik wrote:
Have you even _tried_ to create your dialplan?
And to make it worse, he copied this drivel to the Developers lists.
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Yahoo IM: [EMAIL PROTECTED
it if that could work some how to use those
to select which extension to dial direct to voicemail while on a call, but call
the extension while not on a call.
Any ideas??? THANK YOU!!!
Chris
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I have a client's installation that requires 4 lines PSTN interface only
so I am looking at 4 port FXO units. What works well with Asterisk and
is not exorbitant to purchase? Would a Sangoma remora be better?
Chris Mason
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are running sccp through chan_sccp if that makes any difference
to operation.
Thanks in advance folks.
Regards,
Chris
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replied to me since
--
Chris
- Original Message -
From: Chris Earle (CBL)
To: Areski
Sent: Tuesday, June 13, 2006 6:15 PM
Subject: Re: CDR-Analyser version question
Thank you for the reply;
I see now that the main file cdr.php does work with argument ?s=1, 2,
etc
but when s=0
Thanks for the reply,
--
Chris
- Original Message -
From: Mojo with Horan Company, LLC [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Monday, June 26, 2006 1:49 PM
Subject: Re
for.. (mainly
because again, the person may not hear/know the other phones are ringing).
Thank you for any help!
Chris
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Kevin Smith wrote:
It is making me lean that way, because other phones (same settings)
are using the AC adapters in another office. The ones on the adapter
have not been having this problem, but they don't use the phone much
so they may have never noticed if it did.
If you go into the ftp
Kevin Smith wrote:
Any other thoughts as to what may have caused the phone to reboot?
the power supplies on these phones are very underrated and any power
fluctuation will cause them to reboot. I get it when we are on generator
and the A/C cuts in.
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(264) 497-5670 Fax: (264
readers?
Regards,
Chris
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/ Zaptel has been broken in the update?. Any Ideas? I really need this update as it appears to fix a bug that I have with IAX2 which stops listening to frames and timeouts all calls even when the latency is averaging 5ms.
Thanks
Chris Teesdale
I.T. Development / I.P Telephony Development
.
Any help would be much appreciated!
Regards,
Chris Jones // Network Administrator
Top Level Internet
e: [EMAIL PROTECTED]
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(and, so it would seem,
at various other times too). This has only been occuring since mISDN was
installed.
Any ideas?
Thanks,
Chris
Message: 4
Date: Fri, 9 Jun 2006 18:39:36 +1000
From: MBIT Technologies [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk, mISDN and a Fritz card
To: 'Asterisk
uname -aLinux pbx 2.6.15-1-686 #2 Mon Mar
6 15:27:08 UTC 2006 i686 GNU/Linux
I'm using the 2.6.15-1-686 kernel package from
debian testing. mISDN was installed using the latest CVS
sources.
Regards,
Chris
---
From: "MBIT Technologies" [EMAIL PROTECTED]Subject: RE:
[Aste
holder?
Instead of discussing dongles and other methods of locking down licences to
hardware, how about a discussion about why the lockdown is there in the
first place?
Regards,
Chris
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3 seconds (given that that would result in
ringing for 6 seconds)?
Regards,
Chris
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Asterisk
of how you want to deploy them.
In short, the method of enforcement is poor and leads to resentment from
customers. Surely Digium can construct a better system?
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Yahoo IM
overwriting any modified sound files etc? Should I delete the
current files or move / make a copy to a different location first?
I know this is a lot of questions but I am hoping for a best
practice idea etc
Regards
Chris
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function, I do not hear the voice.
What can cause that? I have not seen this problem before on an Asterisk
system.
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, and wasnt sure if that was
the ONLY way to go.
Thanks!
Chris
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voicemail box or will it have to be a general
one) and 2) do you have the commands I need to enter to tell the Panasonic
system about the * box?
Thank you again!!!
Chris
===
Date: Tue, 30 May 2006 13:36:23 -0400
From: C F [EMAIL PROTECTED
Hey all,
I am not sure if this is the correct place to do this, however, I have
been working on a New Zealand style voice prompt set.
(This has also been announced on New Zealand Asterisk users list)
Finally these have got to a point where I consider them to be stable
and able to be used in a
.
Chris
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either
is a complete set, which means from time to time you still get Allison (USA)
chirping in with the odd sound or two.
Regards,
Chris
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On Tue, 16 May 2006, Tim Sharp [EMAIL PROTECTED] wrote
Chris,
When I tried background it waited until the message was done before
dialing, just like playback. Am I missing something?
Wasn't my suggestion :)
If I've understood what you're trying to I would go one of two ways:
Rather than
=0x8d045333 rev=0x00
hdr=0x00
vendor = 'S3 Graphics Co., Ltd.'
device = '86C420 ProSavage DDR'
class= display
subclass = VGA
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On Fri, 19 May 2006, Dinesh Nair [EMAIL PROTECTED] wrote:
On 05/19/06 16:30 Chris Hastie said the following:
I've just received an OEM Wildcard X100P FXO card. Installing into
my FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at
all. Since
have you downloaded, compiled
On Fri, 19 May 2006, Dinesh Nair [EMAIL PROTECTED] wrote
On 05/19/06 18:57 Chris Hastie said the following:
Yes, I have these. The modules load, but ztcfg complains
ZT_CHANCONFIG failed on channel 1: No such device or address (6) and as I
said, it doesn't appear that the card has been
,Dial(${FOO},20,tm)
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. So if you loose those
rogue commas, the answer to your what's the difference question is a
specific time out value, the ability for both parties to perform
transfers and a ringing tone.
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working if only because I'd like to change
the difficult to maintain
exten = s,3,GoToIf($[${SIPDOMAIN} : ${LOCALREGEX}]?4:20)
in my dial plan to something like
exten = s,3,GoToIf($[${CHECKSIPDOMAIN(${SIPDOMAIN})} = ]?4:20)
I'm using Asterisk 1.2.7.1 on FreeBSD 5.4.
Thanks
--
Chris Hastie
.
Is this a chan_misdn problem or is it a card problem?
We have a number of sites running from 1-3 HFC-based cards in a machine, and
none of them have any significant echo at all. All ours are running with
zaphfc (part of the bristuff package). Might be worth giving that a try.
Regards,
Chris
--
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that. SIP/teliax/${EXTEN} works with teliax
but not voxee.
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boxes that'd fit the
bill, contact me off-list if you want to discuss it further/shameless plug
Regards,
Chris
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This is a known problem and it does not matter what zaptel timer you use. A
solution is available in 'svn head' by using
asterisk.conf
internal_timing = yes
OR
Enable internal timing support (-I)
on the command line. I don't know if this has been backported to the stable
branch.
Chris
Hi List,
Is it possible to store meetme config in a MySQL table?
If so, any pointers would be appreciated.
Thanks
Chris
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reasonably take it? 10 minutes? 1 minute?
Thanks again.
Regards,
Chris
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I have left this a few days but I still can't compile ael_lex.c in HEAD on
CENTOS. I've installed ncurses and bison but I get the following error
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3-O6 -march=i686 -fomit-frame-pointer
-include
phone. But having done that, further inbound SIP calls do
nothing at all - no apparent response from *. All the inbound calls are using
GSM and there is plenty of bandwidth left. It just seems like I can get to five
SIP connections and no more.
--
Chris Hastie
(bearing
in mind I have no control over the ADSL connections the users are subscribed
to)?
Thanks in advance.
Regards,
Chris
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(and are paying BT for the privilege), you may not
wish to go down this route.
Regards,
Chris
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, and I've not had any complaints for the last month since
their box was upgraded to 1.2.
Hope that helps.
(shameless plug - I have a spare TDM400 card here if you're looking to
acquire one at a reduced price - discuss off-list if interested)
Regards,
Chris
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Joshua Colp wrote:
Please don't do product announcements on asterisk-users, that's what
asterisk-biz is for. Thanks!
Please don't send the whole announcement back to the list just to add a
line tot he bottom. Trim.
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Chris Mason
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no responsibility to the
customer and will not bother to tell us they might not be around next
week. Once bitten...
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referenced.
Also, the NTP server must be synchronized before it can be used as a
timing - try ntpq -p to see if there is an asterisk besides a real ntp
server.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
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Cell: 264-235
, server is P4 2.4 GHz.
Monitoring the ADLS does not show any significant packet loss.
Watching the CLI does not show any events, the calls just end.
I am at a loss, what can I do to debug this?
--
Chris Mason
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Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; format=flowed; charset=iso-8859-1;
reply-type=original
- Original Message -
From: Chris Gamble [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 24, 2006
We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are
getting frequent restarts on the spans which lead to dropped calls. I have
pasted some hopefully pertinent information below -- anyone have any clues that
might help?
Thanks
Next line is repeated throughout messages,
*CLI show version
Asterisk 1.2.5 built by root @ host.dyndns.org on a i686 running Linux
on 2006-03-15 19:21:51 UTC
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is of no use for billing. I haven't
found an application that bills from the CDRs, everything I found wanted
to create the database entries. I think ASTPP can read your CDR, though.
--
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NetConcepts
(264) 497-5670 Fax: (264) 497-8463
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Tom Vile wrote:
I am open to suggestions as well.
On 4/18/06, Seth Remington [EMAIL PROTECTED] wrote:
Why is it a problem. I have 800 numbers through Teliax without any problems.
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Chris Mason
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seeing stuff about J1 vs. T1/E1
so does that mean I can't use a Digium card it there?
Can someone please clarify what sort of system I'm looking at here and if I
need a japanese retailer for the card or what
;-)
Thanks!
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Chris Earle
System Solutions Specialist
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This message has been
So it could be BRI, PRI or maybe even Analog there??
I guess what I'm asking is it predominantly ISDN there or not
Thanks for the input about the card and chan-capi
:-)
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Chris
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
To: Chris Earle (CBL) [EMAIL PROTECTED
running psql 7.4.8 on CENTOS.
Chris
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Chris Mason
NetConcepts
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Any ideas?
Just out of curiosity did you happen to put an Answer() before playing
audio or ringing? I use BroadVoice also and I used to have the exact
same problem but putting Answer() as the first step in the context
before playing my menu solved the problem.
-Chris
--
Chris Shaw
-- totally know about the module load order thing and ztcfg -- no
worries there
I've been able to dial out and everything from the start ! -- which is a
bridge from digium--junghanns there..but incoming calls seem to be a
whole other issue. :-(
Exhausted from trying a million things,
Chris
. Been trying everything for weeks
:-(
Chris
Chris Earle (CBL) [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi all,
is it possible to bridge a call between a Junghanns quadBRI card and a
TDM400 in the same server?
It should be I think, -- I am trying this and when an incoming
. please tell me this will work another way
Any comments appreciated, anything!
Chris Earle [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
This is getting very annoying.
Thought it might be a irq conflict/sharing issue -- so resolved that.
Still, cannot get an incoming call
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