Strangely, I can DELETE the messages from the web interface...just playback
causes a permission error...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical
Support
Sent: January 30, 2009 9:40 PM
To: Asterisk Users List
, 2009 at 9:53 PM, OCG Technical Support supp...@ocg.ca
wrote:
Strangely, I can DELETE the messages from the web interface...just playback
causes a permission error...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical
Connect an amp to the onboard speaker (run wires out the case)...then you
can really blast the ring!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: January 28, 2009 10:53 AM
To: Asterisk Users List
Subject: Re:
I've noticed on a few installations that the very first audio played after a
call in answered (eg: Greeting), the first part of the audio is
cutoff/stuttered.
Is this because Asterisk needs some RTP to create a sync for audio - and the
first 1 second is lost? Should one play 1 sec of silence
I'm looking for some ideas of people who have setup a sidecar (eg: Aastra
560M).
Obviously it's handy for BLF (to see who's on a call)...but what else?
Anyone want to share interesting things they've done with a sidecar?
___
-- Bandwidth
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: January 18, 2009 3:11 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Using a sidecar? Ideas?
Importance: High
OCG Technical Support schrieb:
I'm looking
Sent: January 18, 2009 9:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Using a sidecar? Ideas?
Importance: High
Here is a list of stuff that I can remember:
BLF
Group login/logout
Day/Night mode
Call Record
Speed dial
On Sun, Jan 18, 2009 at 2:43 PM, OCG Technical Support supp
If you want to email me your fixed script I'll put it up on the web site...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: January 15, 2009 7:08 PM
To: Asterisk Users List
Subject: Re:
Start with your mail log. Any errors visible?
How about system log - PAMpermission errors?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: January 14, 2009 5:31 PM
To: Asterisk Users List
I can do a great Colonel Klink and pretty good Shulz imitation...in case you
want me to record some prompts.
:)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion
Sent: January 7, 2009 11:31 AM
I see a variety of DECT 6 phones available CHEAP at costco. Is there a way
to convert these to SIP?
I recall someone talking about a Siemens devices that works with all DECT
phones, making them SIP (I think)
___
-- Bandwidth and
: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't
match request NOTIFY to call
Importance: High
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: Wednesday, December 03, 2008 11:14 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk
I'm using the Wm6 built in client. (Enabled via CAB file to add-back files
removed from ROM)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hakem Ta
Sent: December 3, 2008 8:42 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host
mean Windows Live Messenger?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: Wednesday, December 03, 2008 9:15 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't
match request NOTIFY to call
I'm trying to get my Windows Mobile 6 phone working as an asterisk client.
Overall things are working well. However, I regularly get the following
message:
[Nov 27 21:57:28] WARNING[4507]: chan_sip.c:12892 handle_response: Remote
host can't match request NOTIFY to call
'[EMAIL PROTECTED]'.
I've got a user with Linksys ATA's for their analog phones. At random times
during calls, the other party hears DTMF tones during the call.
Is there a way to solve this?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Take a look at smartCID at www.generationd.com
This tool will set callerid based on number in a database. If not listed
there, it will search 411 for reverse lookup etc.
It will also let you flag calls for blocking, etc..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Could you send a link to the post you referenced? I'd like to get sendmail
working with rogers too...
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: November 8, 2008 8:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users]
This was so interesting I had to move it to its own thread!
Is anyone using this script? How does it perform compared to the older
WonderShaper script?
-M-
==
Thanks Kristian I will checkout the new script and see how it goes!
Jonn
-Original
Post it on the wiki! Im sure Ill need it someday
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of César García
Sent: October 29, 2008 6:54 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] XML Cisco config file
Well guys I got it, I started up again making the xml
How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1
out? Etc...
Are you looking for lost RTP between * and internal phones or * and external
provider?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: October 24,
that
picks up calls from the analog lines and routes them to the appropriate
phone, although it will eventually be linked to a larger system once all
the minor bugs are resolved.
OCG Technical Support wrote:
How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1
out? Etc
I have a new Fedora 9 firewall I am setting up in front of an Asterisk 1.4
box. I ported over all of my iptables rules..but now have a strange
problem: SOMETIMES, the audio is only 1-way (i.e. and RTP path problem).
Can someone offer a tip here? Since I have conntrack_sip loaded on the
I have a 7921 wireless phone working with Asterisk, and I want to tighten
the wide open port range of my IPTABLES now.
I tried allowing only SCCP port (2000) in/out and found that my audio was
gone. A quick look at my iptables message shows source port 15886 and dest
port 25968 used:
FORWARD
I'm setting up a 7921 and now want to add a second line to the phone. In my
SCCP.conf file I have:
autologin = 235,299
However, on reloading SCCP the phone fails to login to the second line with
this error:
[Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register:
I've added lines like this:
speeddial = 123,test
speeddial = 260,Bob
in the [device] section for my 7921, but the speed dials do NOT appear on
the menu (click right from the main screen). Am I missing something obvious
here?
Thanks
MD
on Cisco 7921
On 13:15, Fri 12 Sep 08, OCG Technical Support wrote:
I've added lines like this:
speeddial = 123,test
speeddial = 260,Bob
in the [device] section for my 7921, but the speed dials do NOT appear on
the menu (click right from the main screen). Am I missing something
I would have said the short answer is IAX
:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: September 12, 2008 7:31 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Which internet phone protocol best to choose
The short
This is some pretty basic stuff... (someone will probably send you a RTFM)
Start with the sample dialplan (make samples I think)...trace the dialplan
along to understand how it works
Check the wiki and then post anything that you need help with
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I little more digging and I confirmed that cell phone VM and FAX waiting
icons are in fact controlled by a proprietary SMS message format. Here's
what I found:
Message Waiting Indication Group: Store Message
This Group allows an indication to be provided to the user about the status
of
We have a number of clients who have replaced their cell carrier voicemail
with Asterisk (call forward no answer to * box).
One feature they miss is that the cell carriers send the phone a message
showing # voicemails waiting. Can Asterisk do the same somehow?
MD
Technical Support [EMAIL PROTECTED]
wrote:
We have a number of clients who have replaced their cell carrier voicemail
with Asterisk (call forward no answer to * box).
One feature they miss is that the cell carriers send the phone a message
showing # voicemails waiting. Can Asterisk do the same
On Sun, Jun 22, 2008 at 11:23 AM, OCG Technical Support [EMAIL PROTECTED]
wrote:
We have a number of clients who have replaced their cell carrier
voicemail
with Asterisk (call forward no answer to * box).
One feature they miss is that the cell carriers send the phone a
message
showing
-0400, OCG Technical Support wrote:
I've tried a few approaches to making the multimedia keys on my kbd play
nice with myth, but all have lead to dead ends.
One such dead end is to post this question to the Asteris Users mailing
list, I guess :-(
Wrong list?
--
Tzafrir Cohen
icq
I've tried a few approaches to making the multimedia keys on my kbd play
nice with myth, but all have lead to dead ends.
I decided to take the simple approach, and use the myth setup menu for
keyboard mappings. Now, I have myth (0.20) waiting for a key with Press a
key, but when I press the
I'm looking at building up a standard asterisk system fanless/no moving
parts. I found a cheap solid state disk (Transcend TS32GSSD25S-M), but it
is SLOW...25mb/sec read 8mb/sec write.
Has anyone tried a slow disk like this on asterisk? Will this delay voice
prompts or screw up ast/linux in
Change the order of resolution (hosts first, then DNS) and add relevant
entries to your hosts table. That makes asterisk happy w/o an internet
connection.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: June 9, 2008 9:09 PM
To:
Has anyone create the necessary config/kbd file to allow the DiNovo mini to
work well with myth? (Mapped all of the multimedia buttons etc)
=MD=
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
PROTECTED] On Behalf Of OCG Technical
Support
Sent: June 6, 2008 9:03 PM
To: Asterisk Users List
Subject: [asterisk-users] Logitech DiNovo Mini keyboard with myth
Has anyone create the necessary config/kbd file to allow the DiNovo mini to
work well with myth? (Mapped all of the multimedia buttons etc
Anyone tried Asterisk with Fedora 9 (recent release)?
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Permissions? Try running msmtp from the asterisk account? (Assuming that
is how you have it setup)
I don't know msmtp - but is there a maillog equivalent?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roberto Milani
Sent: May 13, 2008 7:49 PM
We also have a script available (on www.generationd.com) which allows a user
to reply to an emailed voicemail, which then deletes the associated VM file
on the asterisk box.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent:
A client has asked that our asterisk installation leverage their large
investment in their existing data center infrastructure. We're thinking
about putting the voicemail messages onto a Samba share (on their file
servers). Any pros/cons to this? Does network/samba latency create
choppiness?
Like many users I get my voicemails emailed to me, AND left on the asterisk
server, so that I can retrieve them by phone or by email. However, I was
frustrated that after I deleted a message in outlook that I still had to
delete it from asterisk manually.
So, I wrote a script that runs on the
After lots of interest I've stopped emailing people the script and have made
it available for download from www.generationd.com Look in the Downloads |
Asterisk section.
Be sure to read the readme AND the top of the script for instructions...
___
--
Another useful script for those interested
On the www.generationd.com web site you will now find the asteriskcontrol
script file. This script can automatically restart Asterisk (gracefully)
following a change in external IP address - for dynamic IP hosts. As well,
it can update the
We did a custom Goldmine customer lookup popup based on Asterisk CID...but
that's about it.
What are you trying to do?
MD
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: March-24-08 9:05 AM
To: Asterisk Users List
Subject: [asterisk-users] Gold Mine CRM +
Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in
the voicemail messaging emailing operation. I had set serveremail option
to:
[EMAIL PROTECTED]
and under ast 1.2.x messages arrived at user mailboxes from
[EMAIL PROTECTED] . However, since upgrading emails
PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:41 AM
To: Asterisk Users List
Subject: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18
Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in
the voicemail messaging emailing operation. I
to tell sendmail to trust the asterisk account or
voicemail from address
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:57 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Bug in voicemail's serveremail setting in
1.4.18
RESOLVED! For others fiting a similar problem look at
/etc/mail/service.switch
This is the only way to force sendmail to not do a DNS lookup (first)...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:57 AM
To: Asterisk Users
I (like many others probably have) added the sender of the invite to my spam
filter. That avoids the many replies - and also blocks future email from
someone stupid enough to spam multiple entire list with an invite!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I have an unusual and recurring problem since I upgraded to Asterisk 1.4.
Sometimes, mid-way through a call, I hear 6 shorts beeps and then the
inbound voice quality degrades massively. It sounds like the other user is
a robot...etc.
I'm guessing something (aastra 480 or Asterisk 1.4) is warning
We are installing Aastra phones (480's and 57i's) into a fairly simple
asterisk setup. Although call park pickup work fine using xfer to 700 (to
park), dial 701 (to pickup), we are unable to make the park/pickup softkey
feature work on the aastra's.
Although we've programmed the softkeys per
Are the 7921G phones convertable to SIP too?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sigma Networks
Sent: March-01-08 11:56 AM
To: Asterisk Users List
Subject: [asterisk-users] Cisco 79xx users/consultants, 7970G color in
particular share
.
On Mon, Feb 25, 2008 at 9:41 PM, OCG Technical Support
[EMAIL PROTECTED] wrote:
I'm still struggling to pickup calls. I now have a single context
(entryocginternal) where I have include = parkedcalls.
The log below shows me calling from one internal extension
to another
It looks like I have a conflict! (See results of diaplan show below). How
can I force the parkedcalls context to be matched first? (I include
parkedcalls before I define the _X. priority).
pbx*CLI dialplan show [EMAIL PROTECTED]
[ Context 'entryocginternal' created by 'pbx_config' ]
'_X.' =
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
OCG Technical Support
Sent: Tuesday, February 26, 2008 10:02 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Parked calls - can't pickup
It looks like I have a conflict! (See results of diaplan
show below
I'm still struggling to pickup calls. I now have a single context
(entryocginternal) where I have include = parkedcalls.
The log below shows me calling from one internal extension to another, then
picking up, then parking the call.
-- SIP/239-0915d5c8 is ringing
-- SIP/239-0915d5c8
Can someone point me
to call center reports available from Asterisk? We setup a small call
center with agents, and will now be looking at reports.
Ideas?
Thanks,
MD
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
Look into NVDETECT, and fax2mail script on www.generationd.com
Fax detection is automatic
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Monday, September 04, 2006 5:05 PM
To: asterisk-users@lists.digium.com
Subject: Re:
We have a client
running Asterisk using a dynamic IP. When the IP lease is renewed
toa different address, their SIP connections to external clients fail (one
way audio).
A simple asterisk
restart fixes the problem, but they can't count on an admin always
available. Is there way in
We use NV's fax detection and it works very well.
(However this can still congest your system with junk
faxes).
We combined this with cid_rewrite (from www.generationd.com) which allows blocking
of calls based on CID name/num (and also rewrites the CID name based on your own
SQL database).
I frequently find voice messages are emailed to users with insufficient
volume - barely audible. I would like to have asterisk run a sox command to
adjust the volume of each message before emailing (perhaps once the message
has been left).
Has anyone done this? Care to share the steps?
Thanks,
Why not capture the faxes (in or out) in tiff format, instead of audio
format? Setup your asterisk box to relay faxes!
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Thursday, May 04, 2006 10:07 AM
To: Asterisk Users Mailing
Interesting question! If you have the audio in only
(assuming it was a fax received) or audio out only (assuming it was a fax sent),
and you pair with an identical fax machine to the original (assuming it responds
exactly the same in terms of handshakes, speeds, ECM, etc) then it might
Check your voicemail.conf and sip.conf - I suspect that you have multiple
mailboxes and they are not associated with the right SIP device.
MS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tofik
Suleymanov
Sent: Saturday, April 15, 2006 2:48 PM
To:
Matt:
Some great scripts on your site! I read the one about blacklisting a
callerID. As an alternative (albeit more complex), we have a posted a
utility which looks up caller phone numbers in a mySQL database and takes
action based on one field of the database (Drop it, say busy, or let caller
That's the nature of consulting - you have to balance
demonstrating competency with solving the problem before being paid. We've
had many similar experiences, and we now require prepayment for 2 hrs service
before we do any work (or even talk to the client for more than a
fewminutes).
Miles,
I think this is a limitation of the AGI - I don't believe that asterisk can
fork a new process. If so, that would be interesting!
The script uses Wget - I believe we can set a timeout so that your system
doesn't hang waiting for the HTTP response. Let me know if that would solve
your
Jay,
I contacted you many times regarding the script, whether you planned to
update it, suggestions for features, etc. You did not respond to any of my
later emails. Similarly, there was discussion between list members
regarding whether this script was orphaned after changes to 411.com made the
Miles,
You can also download cid_rewrite from www.generationd.com This PHP script
looks up the phone numbers in a local MySQL table, and/or uses reverse 411
on the web to lookup the name, and/or more options.
Michelle
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
You can't reliably run a real-time application (like
asterisk) on a virtual machine. You will get better performance from an
old PC than a VM on a new top-end PC. Sorry
MD
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Tuesday, March 28, 2006 12:40
line is that PC virtualization and real-time applications don't
mix. You are better off with less horsepower without latency issues.
MD
-Original Message-
From: Greg Oliver [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 28, 2006 1:19 PM
To: Technical Support; Asterisk Users Mailing List
James,
Which fax application are you
using?
MD
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
HawksSent: Monday, March 27, 2006 5:29 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] txfax
problem
I can not get txfax to work with a multiple page
Try sending email from the command prompt
first...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
hugolivudeSent: Friday, March 17, 2006 10:06 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] problems with emailing voicemail
Hi,
I'm
We contractedlocal voice talents to handle IVR
recordings (in male female, French English). You're right that
it doesn't match the system prompts, but in some cases we want that when
switching to our IVRapplication.
MD
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zach
Try pressing FLASH, then *1 and then FLASH again.
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
We are working on a smartDND agi script which will do this. Should be
coming out this spring :)
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, February 27, 2006 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
We've setup some FC3 systems with T1 at 100% - no problems. As noted below,
I suspect other processes (or lack of computing power) may be causing your
issues. Even things like turning off the gui (run level 3) can make a big
improvement.
MD
-Original Message-
From: [EMAIL PROTECTED]
is shared with other apps, you may
not be getting the time slices you need (very different from overall CPU power
you are getting).
Can you get onto your own box at the
ASP?
MD
From: J Poz [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 18, 2006 11:35 PMTo: Technical
Support; Asterisk
This seems pretty commercial for a non-commercial list!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Sunday, February 19, 2006 10:33 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Intro to
J:
We developed the mail2fax application (www.generationd.com) - so we should be
able to give some insight. I think you are confusing the time to "process"
the incoming (by email) fax document, and the time to fax the document.
Fax over IP causes an enormous number of retries - thus delays.
Try turning off iptables (firewall)
service.
MD
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abhimanyu
RapriaSent: Friday, February 17, 2006 2:19 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP Problem
Fedora Core 4 and Asterisk 1.2.4
Fedora:Linux
We've setup automatic printing of faxes from asterisk (spool to a net queue
or direct to local printer), use the sendmail capability of emailing to an
executable (alias). Why send an image over to asterisk from the TNT?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I think that some people try to make their asterisk box a do-everything
super server. Can you image a traditional PBX with direct access via the
internet, serving web pages via apache, running sendmail, etc.
Our approach has been keep it simple. We lock each Asterisk PBX down has
hard as
easy. Yes, I took the easy way out!
Of course, in my situation I make sure to keep it fairly up to date.
Joseph Tanner
Technical Support wrote:
I think that some people try to make their asterisk box a
do-everything super server. Can you image a traditional PBX with
direct access via
There's a vox forum that focuses on Sipuras - post your query there for good
tech help. We've deployed a number of Sipura's and haven't experienced that
problem (yet). Have you started with the basics: firmware version, analog
cabling, etc.
MD
-Original Message-
From: [EMAIL
We run FC4 on our production installs. It runs great. I should caution you
that just because an update is available, it doesn't mean you SHOULD update.
Treat your FC4 install as frozen - if it works don't update it!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Is there a web portal available for users to:
1. Change greetings
2. Set DND settings
3. Play messages from the web
Etc.
MD
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options
We have tried 3 different ITSP's in Canada - 2 of which had terrible
performance (choppy voice, etc). We've been with unlimitel.ca for a while
now and had good results. They have Toronto and Ottawa DIDs. They also have
good tech support (small company means everyone knows what's going on, and
We're Ontario based, but gladly work across Canada!
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
At first glance it sounds like a routing issue. Are you IAX to your phones,
but SIP to your tisp provider? Any change on your asterisk box / firewall /
ISP / TISP since then?
My first guess is that when you replaced the card you changed a network /
iptables setting etc. on your asterisk box.
Why would the software halt on that date? Is there a time bomb in Asterisk?
I can't imagine what legit piece of code would be checking for a particular
date
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, January 25,
Take a look at cid_rewrite from www.generationd.com This automates name
lookup (based on reverse phone number) using 411.com, a local database of
callerID's, ability to block users based on a parameter, etc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Downgrade your spandsp. Do some reading on spandsp first!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Allan Gee
Sent: Tuesday, January 24, 2006 10:36 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] txfax application problem
Nobody
may have to pay someone to go onto your system to untangle the mess (or
post your problems and analysis one step at a time).
Michelle
-Original Message-
From: Allan Gee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 24, 2006 12:06 PM
To: Technical Support
Subject: RE: [Asterisk-Users
Check out www.generationd.com for their fax2mail and mail2fax scripts. It
might make life simpler
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, January 22, 2006 9:56 PM
To: Asterisk Users Mailing List - Non-Commercial
there is another problem
this might help)
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
quality.
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
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