You may want to start from the basics:
http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu
I hope this helps
l.
2009/10/17 Nazir Ahmed Vaid nazir.v...@gmail.com
Ladies and Gentlemen,
We already have an Asterisk Call center suite installed at our contact
center. Now we wish to commence
Have you given some consideration to learning just a little bit - not
much, just a little introductory basics - about the product before
asking a question?
The answer to your question can be found in basic dial plan
functionality, unless you need a very complex IVR engine driven by
something
2009/9/29 Alec Davis siva...@paradise.net.nz
When I say no reliable internet, some days it's good, others it's not, so
to
try to push a call over an IAX trunk is going to fail.
For the choice of providers, when it comes to a business with branches
around the country (NZ is small enough)
Hi,
I looked at KVM's changelog (http://www.linux-kvm.org/page/ChangeLog) where
several entries relate to PCI.
Is KVM suitable now for small scale Asterisk (involving using PCI telephony
cards) development workstation ?
By that, I mean :
- install 2 or 3 PCI telephony cards in a Linux platform,
Hi,
I explain what I want to do..
All the operators share their phones. The number of the operator isn't
constant, so it's possible that two operators share all the phones.
They need to move all around, so they pick up the first phone they find.
If there are only few operator is very annoying for
Hi,
I have three Snom M3s at the moment but getting pretty fed up with the issues
:( I am UK based and would be interested to hear of other peoples
recommendations. Key features :-
* VM Notification
* Good Range
* G729 codec support
* Common/Private Address Books per Handset(s)
TIA,
Best
On Sat, 17 Oct 2009, --[ UxBoD ]-- wrote:
Hi,
I have three Snom M3s at the moment but getting pretty fed up with the
issues :( I am UK based and would be interested to hear of other peoples
recommendations. Key features :-
* VM Notification
* Good Range
* G729 codec support
*
- Gordon Henderson gordon+aster...@drogon.net wrote:
| On Sat, 17 Oct 2009, --[ UxBoD ]-- wrote:
|
| Hi,
|
| I have three Snom M3s at the moment but getting pretty fed up with
| the
| issues :( I am UK based and would be interested to hear of other
| peoples
| recommendations. Key
Does anybody know what this message means?
[2009-10-17 08:11:59] DEBUG[9598]: sched.c:204 sched_settime: Request to
schedule in the past?!?!
I've tried all I can find on google with no change
It seems to happen when call is at a READ() or Background() - What the caller
hear is small delays
- --[ UxBoD ]-- ux...@splatnix.net wrote:
| - Gordon Henderson gordon+aster...@drogon.net wrote:
|
| | On Sat, 17 Oct 2009, --[ UxBoD ]-- wrote:
| |
| | Hi,
| |
| | I have three Snom M3s at the moment but getting pretty fed up
| with
| | the
| | issues :( I am UK based and would be
On Sat, Oct 17, 2009 at 7:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
| I have three Snom M3s at the moment but getting pretty fed up with
| the
| issues :( I am UK based and would be interested to hear of other
| peoples
The S685IP has no headset jack AFAIK. If you want to use a headset
Is this patch correct? The doesn't make logical sense to me. I think
it should be || and making this change fixes the problem I have with SIP
phones in MeetMe conferences. If it's correct, is there someplace more
formal that I should submit it to?
*** app_meetme.c.old2009-10-11
Randy R wrote:
On Sat, Oct 17, 2009 at 7:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
| I have three Snom M3s at the moment but getting pretty fed up with
| the
| issues :( I am UK based and would be interested to hear of other
| peoples
The S685IP has no headset jack AFAIK. If you
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner ken...@gnat.com wrote:
Is this patch correct? The doesn't make logical sense to me. I think
it should be || and making this change fixes the problem I have with SIP
phones in MeetMe conferences. If it's correct, is there someplace more
I have used the group function to limit the calls entering a queue for a
similar reason to yourself.
PaulH
Niccolò Belli wrote:
Hi,
I explain what I want to do..
All the operators share their phones. The number of the operator isn't
constant, so it's possible that two operators share all
2009/10/17 Randy R randulo2...@gmail.com:
On Sat, Oct 17, 2009 at 7:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
| I have three Snom M3s at the moment but getting pretty fed up with
| the
| issues :( I am UK based and would be interested to hear of other
| peoples
The S685IP has no
At Astricon 2009, I presented on best practices for reliable carrier
grade telephony. The presentation is available for download from:
http://integrics.com/presentations/astricon_2009.ppt
--
Alistair Cunningham
+1 888 468 3111
+44 20 799 39 799
http://integrics.com/
Hello,
I was wondering if anyone has any insights on the best way to automatically
monitor an asterisk box to check it is constantly available and processing
calls.
Many thanks
Dan
IT Maintenance Contract Clients can now access our Instant Chat
Dan Journo wrote:
Hello,
I was wondering if anyone has any insights on the best way to
automatically monitor an asterisk box to check it is constantly
available and processing calls.
Depends on which signaling technology is in use.
I'll assume SIP here.
Checking whether it's processing
We use Nagios. It is by far the most popular system for making sure process
are running and whatnot..
www.nagios.org
--
Jarrod Lash, jar...@fed-com.com
Federated Communications
Office: +1-412-357-2127
Mobile: +1-412-999-0049
Fax: +1-412-545-8368
On Sat, Oct 17, 2009 at 8:30 PM, Dan Journo
Nagios has a plugin check_sip that can be used for this.
-Jai
On Sat, Oct 17, 2009 at 5:30 PM, Dan Journo
d...@keshercommunications.comwrote:
Hello,
I was wondering if anyone has any insights on the best way to automatically
monitor an asterisk box to check it is constantly available
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner ken...@gnat.com wrote:
Is this patch correct? The doesn't make logical sense to me. I think
it should be || and making this change fixes the problem I have with SIP
phones in MeetMe conferences. If it's correct, is there someplace more
formal
On Thu, Oct 15, 2009 at 12:24 PM, Olivier oza-4...@myamail.com wrote:
Hi,
I've downloaded for a demo, a P0S3-08-12.zip file which is suppose to work
with 7960.
Is it supposed to be the same file that the one needed to 7942 model ?
No. The SIP firmware for each model are different except for
On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner ken...@gnat.com wrote:
Is this patch correct? The doesn't make logical sense to me. I think
it should be || and making this change fixes the problem I have with SIP
phones in MeetMe conferences. If it's correct, is there someplace more
formal
Hi Darrin
Thanks for your kind reply.
Your description is right, PC(Soft Phone) ADSL Router Internet Asterisk
box
Thanks for your suggestion on the security.
Please advise , I am specifically concerned about the port to which server
reply after initial communication (random above 32000)
David Backeberg wrote:
From a quick glance at your patch, I would say that it probably tries
to address the audio quality problems I and others were experiencing.
No, it's fixing a much more serious issue. As I sent to this list twice,
when I have a conference between Dahdi ports and SIP
Wish I could have made it :( Is there a possibility of a collection of
the talks/slides/handouts/videos/presentations for download? Even pay
for?
Cheers,
j
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 -
Most probably. They usually get uploaded some time a little later.
Jeff LaCoursiere wrote:
Wish I could have made it :( Is there a possibility of a collection of
the talks/slides/handouts/videos/presentations for download? Even pay
for?
Cheers,
j
I'm told that they will show up on the event site in about three weeks.
On Sun, 18 Oct 2009 02:29:48 + (UTC), Jeff LaCoursiere wrote:
Wish I could have made it :( Is there a possibility of a collection of
the talks/slides/handouts/videos/presentations for download? Even pay
for?
What version are you running?
1.6.2.0-rc2
Does that version support disabling talker optimization?
Yes.
Have you tried disabling talker optimization?
Yes. That's how I found the bug. I got no audio from the SIP phone
into the conference, so I decided I'd try seeing if it did if the SIP
On Sun, 2009-10-18 at 01:30 +0100, Dan Journo wrote:
Hello,
I was wondering if anyone has any insights on the best way to
automatically monitor an asterisk box to check it is constantly
available and processing calls.
snip
It depends on how deep and far you want to go. As already
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