Asterisk has got to be about the best kept secret in telephony. I've seen
numerous articles on slashdot about VoIP, even in relation to Linux and
only *once* has the post even mentioned Asterisk. Am I missing something,
or is Asterisk clearly a good potential player in any kind of linux-based
Yes, it is a well-kept secret, which is a shame since it obviously
fits so many different requirements. Here are some late-night
musings as to why new users coming to Asterisk is only a stream when
it should be a river:
1) No 1.0 release. In fact, no release structure at all really.
(Hold
I'm new to Asterisk and I completely agree with Mark. Asterisks is the best
kept secret in telephony. I cannot recall how I originally found Asterisk,
but I do remember spending far too much time surfing before I did.
One place I found seems to mention nearly all products, except Asterisk.
Asterisk would need scalability and redundancy on the voip side to
play in the soft-switch area. The biggest issue stopping Asterisk having
redundancy and scalability using sip is the inability to work with just
about any sip device without canreinvite turn off. If Asterisk could
handled
Besides you got list four times since May!smile
http://slashdot.org/search.pl?query=asterisk
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Can I add to this and say that another thing that could be hindering the
takup is Single System VoIP scalability and a certain amout of
Enterprise flexibility..
Let me explain those two..
Before you start reading these and thinking This guy is mad!! let me
just say that I love Asterisk and
You need to add nat=yes
for the sip phones in sip.conf, IMHO
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt
Sent: Saturday, November 08, 2003
1:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk
over VPN.
Hi
People,
Let's
take a
Hi,
Thanks for info,
Didn't know the mails were sent as HTML, will check the email settings.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling
Sent: 08 November 2003 02:03
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Streaming MOH
Is there a way to put a call on hold and play music on hold with out
using the park app?
Yes there is. ;-)
Philipp
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Hi!
Is there any way to grab an audio stream and pipe it out as the MOH?
I am a helper at a local Charity Hospital Radio Station and thought it
would be nice to pipe the studio output to waiting callers.
Look here:
http://bugs.digium.com/bug_view_page.php?bug_id=413
Philipp
Hi!
1) No 1.0 release. In fact, no release structure at all really.
(Hold your flames: I know this is to be remedied soon, along with
backtrack patches for security/stability.)
With that comes a changelog and some basic documentation. I still find
it amazing that coders are permitted to
On Thu, Nov 06, 2003 at 12:57:49AM -0500, William Carlson wrote:
Ok I see the confusion. I actually do have a TFTP server running on the
asterisk machine but it does not have any Skinny stuff just ringtones and
logos for my SIP 7960's. The id is found under settings then model
information just
This is where I got the ringtones.
http://www.loligo.com/asterisk/sounds/
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 08, 2003 9:25 AM
Subject: Re: [Asterisk-Users] Skinny (SCCP) help
On Thu, Nov 06, 2003 at 12:57:49AM
woops I ment
http://www.loligo.com/asterisk/Cisco/79xx/current/
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 08, 2003 9:25 AM
Subject: Re: [Asterisk-Users] Skinny (SCCP) help
On Thu, Nov 06, 2003 at 12:57:49AM -0500,
Engineers of all kinds can be a bit lax about documentation. However,
the documentation police are rightly held in a regard usually reserved
for lawyers, realtors, used car salesmen and serial killers.
There isn't a single thing to stop anyone that really loves
documentation actually producing
The 6.0 image is available for download from Cisco TAC. The 6.0 image does
support auto answer (Intercom.)
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent:
Hi,
- Original Message -
From: Jon Pounder [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 08, 2003 12:30 AM
Subject: [Asterisk-Users] diax request
First of all great job on diax. I downloaded it and tried it, could not
connect, got an authentication rejected,but I
Nice this image lets my flakey 7960 run the SIP software :)
Thanks,
Will
- Original Message -
From: Paul Mahler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 08, 2003 10:09 AM
Subject: RE: [Asterisk-Users] 6.0 image for Cisco 7960's?
The 6.0 image is available
On Fri, 7 Nov 2003 23:50:06 -0800, Darren Martz [EMAIL PROTECTED] wrote:
If I'm out of place in the following suggestions, I'm sure others will tell
me grin
- Create a clean SDK of the wonderful IAX2 protocol for Win32 and Mac to
gain exposure everywhere
- Push, entice, bribe IP phone designers
Is any one else having problems with the Snom 200 MWI??
If flashes and shows me there is a message then I go and listen to the
message but the MWI does not clear.. The only way I have found to clear
the MWI is to reboot the phone..
___
Asterisk has got to be about the best kept secret in telephony. I've seen
numerous articles on slashdot about VoIP, even in relation to Linux and
only *once* has the post even mentioned Asterisk. Am I missing something,
or is Asterisk clearly a good potential player in any kind of
Is any one else having problems with the Snom 200 MWI??
If flashes and shows me there is a message then I go and listen to the
message but the MWI does not clear.. The only way I have found to clear
the MWI is to reboot the phone..
Gus,
Works correct for me. Running v2.02q software on
Asterisk has got to be about the best kept secret in telephony. I've seen
numerous articles on slashdot about VoIP, even in relation to Linux and
only *once* has the post even mentioned Asterisk. Am I missing something,
or is Asterisk clearly a good potential player in any kind of
Works correct for me. Running v2.02q software on the 200, and just finished
testing it.
Thanks I will have to play a little more then.. What date CVS of
Asterisk are you running?
Gus,
CLI show version
Asterisk CVS-10/25/03-13:22:42 built by [EMAIL PROTECTED] on a i686 running Linux
CLI
On Sat, 2003-11-08 at 09:20, Rich Adamson wrote:
John Todd's sample config's have been a good first step for newbies, but
the average newbie doesn't have clue where to find them (as one example
only) until after burning up the list with questions. After internalizing
those configs and then
Look in your mailbox somewhere in /var/spool/asterisk. If there is a
gap in the MSG sequence numbers, or if there's a stray file in there it
will make Asterisk think you have new messages even if you don't have
new messages when you check your voicemail.
On Sat, 2003-11-08 at 12:48, Rich Adamson
I sent this yesterday, but for some reawson it did not go through.
Yes,
ASTERISK1 = 2x TDM400P
ASTERISK2 = 3x X100P
I still cannot get it working past that. Is there something screwey with the
wcfxs drivers and Linux?
- Original Message -
From: Louis-David Mitterrand [EMAIL PROTECTED]
As we know market has thousands of great free source products, but somehow
most of companies are buying commercial software and paying a lot of $$$ .
Question becomes why they need to pay so much money for something what can
be taken for free ? Also, why all these software products are so
Steven Critchfield wrote:
On Fri, 2003-11-07 at 16:04, Olle E. Johansson wrote:
Steven Critchfield wrote:
We have to rename Zaptel timing to Asterisk timer, which is more correct
since there are several ways of getting a timer to work, only one of them
is by using Zaptel cards.
Hi all.
I've made a patch for chan_oss.c to enable
callgroups and pickupgroups in it (since wasn't enabled).
I needed it for a special use of the console (pickup
calls arriving to the console from another phone)
btw, If someone is interested, I can submit a patch
to the bugtracker. I won't do it
I beleive there's an (at least below) unmentioned argument why Asterisk
may fail getting really big:
Since Digium/Mark refuses to include any code that isn't copyrighted
Digium/Mark, I'm afraid quite a few developers may be effectively
excluded. By requiring a 'giveaway' to Digium/Mark, we
Take a look at iaxclient.sourceforge.net
The current CVS version supports IAX or IAX2, and works on Win32,
ia386Linux and
Macs.
There are also a few working crossplatform softphones there.
...and iaxclient is probably not one of them. Working softphones for me
includes stability, intuitive
I've stopped referring people
directly to my Asterisk site and instead refer them to the Unofficial
Links page at Digium.
--Eric
--
Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/
;-)
/O
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So what do people think about adding the call rate to the CDR
structure??
This would allow you to detail a call with the rate that was
in affect for that call. When you come back later and do
the billing for the customer you would have the actual per min
rate in the record.
I think this solves
I actually meant on IRC. I had forgotten about my .sig, LOL! Thanks
for pointing it out.
--Eric
On Sat, 2003-11-08 at 14:59, Olle E. Johansson wrote:
I've stopped referring people
directly to my Asterisk site and instead refer them to the Unofficial
Links page at Digium.
--Eric
So what do people think about adding the call rate to the CDR
structure??
This would allow you to detail a call with the rate that was
in affect for that call. When you come back later and do
the billing for the customer you would have the actual per min
rate in the record.
I think this solves an
I personally think the right place to deal with billing is in your
billing application. Your billing application should know about special
rates for different customers, time of day rates, destination rates,
etc. There is already enough information in the CDR to know which
provider you are going
At 01:08 PM 11/8/2003, you wrote:
So what do people think about adding the call rate to the CDR
structure??
Sounds great, but there's one problem. How does asterisk know what the
current rate in effect is? I can think of several ways to do this, but they
all involve some fairly significant C
He did write the new one where you could append say value1:value2 into
that field. Still not pretty but functional.
bkw
On Sat, 8 Nov 2003, John Todd wrote:
So what do people think about adding the call rate to the CDR
structure??
This would allow you to detail a call with the rate that
What is the best way in getting started evaluating Asterisk? Are there
recommendations on the types of card I should be using for an initial eval?
Thanks
Peter
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As rates change of the course of a month, it would be nice to
know what the rate was at the time of the call.
some detail records :)
On Sat, Nov 08, 2003 at 03:37:13PM -0600, Eric Wieling wrote:
I personally think the right place to deal with billing is in your
billing application. Your
download the code,
complile the code,
start bashing on configs. :)
if you want to glue to the PSTN, i'd
recommend getting a FXO (WC-X100P) and FXS (TDM-10B)
card and some cheap SIP / VoIP phones (grandstream or snom)
john brown, ceo
chagres technologies, inc
Providers of VoIP hardware
On Sat, 8 Nov 2003 21:59:43 +0100, Roy Sigurd Karlsbakk [EMAIL PROTECTED]
wrote:
Take a look at iaxclient.sourceforge.net
The current CVS version supports IAX or IAX2, and works on Win32,
ia386Linux and
Macs.
There are also a few working crossplatform softphones there.
...and iaxclient
Sorry to do this to the list, but I have no choice .
Walker,
I've been trying to send you an email off-list for the last couple of weeks,
but one of my mail-hops is failing, do you have alternative address that I
can try ???
Paul
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Asterisk-Users
On Sat, 8 Nov 2003 15:30:14 -0700, John Brown (CV)
[EMAIL PROTECTED] wrote:
As rates change of the course of a month, it would be nice to
know what the rate was at the time of the call.
some detail records :)
Then tell your billing package that, effective at 5:23pm on October 24,
the price of a
I think it is time to start commercial Pro version (not expensive !!!) of
Asterisk.
In my company we already made decision to do it, to offer people
ready-to-go solution. But is is hard to do anykind of such product without
Digium and Mark's support.
Mark I think you are very overloaded
Ryan Tucker wrote:
Call rating is a relatively complex task, traditionally done by people
over in the business office. Stuffing rating code into Asterisk is
not going to be all that useful for all but the most simple
applications. Our billing is complex enough that we outsource it to a
Robert,
You are right about licensing issues. But I have mentioned in my email that
we need direct support from Mark and Digium.
It might be direct license or something else. But we need Digium and Mark
to participate in this project.
Now about commercialization : Idea is to create enhanced
I figured out what was going on with the lack of/stuck on stuttered dial
tone. Apparently, there are two voicemail directories being referenced:
/var/spool/asterisk/voicemail/default, and
/var/spool/asterisk/voicemail/local. The sip phones were using
/var/spool/asterisk/voicemail/local to
I've seen that myself - both on Snom 100 and Snom 200 devices using the
latest beta firmware. I did however suspect this was a bug in the Snom
devices. Generally whatever comes out of the speaker sounds crappy - even
ringing sometimes. Also it seems to come an go and wasn't like this a
couple
Hi Everybody,
Has anybody tried the above (or indeed any other 4XBRI cards) successfully
with Asterisk. As far as I can see the above mentioned card is an active
ISDN card but supported by it's own I4L driver. This leads to interesting
questions particularly regarding echo cancellations (which
I figured out what was going on with the lack of/stuck on stuttered dial
tone. Apparently, there are two voicemail directories being referenced:
/var/spool/asterisk/voicemail/default, and
/var/spool/asterisk/voicemail/local. The sip phones were using
/var/spool/asterisk/voicemail/local to
Don't know if you seen my earlier post, but my 200 has been working just
great since v2.02q was installed a couple of days ago. Speakerphone sounds
fine, MWI works fine, etc. This release seems to be the best I've seen for
a while.
Rich
I've seen that myself - both on
I have been getting disconnects on my pots lines from bellsouth recently. The BS
repairman determined that their was an audible 60 HZ hum on all of my fxo ports. We
also measured the following impedances between tip and ring:
T-R 35K ohms
R-R 90K ohms
T-T 16K ohms
The two latter measurements
John,
The only thing that should be in the call data record (CDR) is data
about the call. You want to do all your rating of calls in the billing
system. That way you offload the additional processing to a batch
activity.
You have enough data in the CDR to combine it with
Well - problem with this fast moving technology. No - my last update was
about 10 days old. My phones are downloading as I write this :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: 09 November 2003 10:34
To: [EMAIL PROTECTED]
And it seemed to have moved even faster than I imagined. My phone is now at
2.02r and not q. Wonder how that's going to work. I might have missed the
only working version without even having had the pleasure of trying it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Solution: The context used in voicemail.conf has to match the default context
in sip.conf.
Sip.conf:
[general]
port=5060
bindaddr=192.168.17.2
tos=lowdelay
disallow=all
allow=ulaw
context=default ; Note: this must match voicemail.conf
;
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