: Sherwood McGowan [EMAIL PROTECTED]
- Subject: RE: [Asterisk-Users] Asterisk 1.2.0-beta2 Released
- Date: Tue, 1 Nov 2005 08:45:52 -0500
- To: 'Asterisk Users Mailing List - Non-Commercial
-Discussion' asterisk-users@lists.digium.com
-
-
- I agree, I would definitely love to find out more about a lot
I'm STILL waiting on an answer to my exact same question. That was around 3
weeks ago
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-John Biundo
-Sent: Tuesday, November 01, 2005 1:13 PM
-To: asterisk-users@lists.digium.com
-Subject:
You could always just add some
exten =
NUM,PRIO,VERBOSE(LEVEL|${DATETIME} -- THIS EVENT HAPPENED)
type commands in your dialplan to force output of the date
time, and you can even reduce the amount of verbosity to the CLI by using it
liberally to signify events, so you don't have to
Just saw this thread.. Wanted to know if you'd like some input from me...
I'm developing ARTCP for controlling, managing, and end-user access to
Asterisk RealTime
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-astgroups
-Sent: Wednesday, October
As part of my overall project, I'm working on some PHP scripts that will do
just that.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Chris Bagnall
-Sent: Friday, October 28, 2005 6:08 AM
-To: 'Asterisk Users Mailing List - Non-Commercial
You could always (I'll actually do it, I have similar scripts written) just
whip up a php script that connects to a Asterisk Manager Proxy (to limit the
possibility of crashing the server by making too many Manager API
connections), and have it issue the following commands:
Action: Command
I've been given the charge of finding out if anyone has gotten vmail.cgi to
work with asterisk realtime, pulling the voicemail users from the db...
I thank you all for any input you may have
Sherwood
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I agree... I've got wy to many customers out there who are pissed
because they thought VOIP would be just as reliable (or even close) as POTS.
SKM
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-[EMAIL PROTECTED]
-Sent: Thursday, October 27,
Sorry to do this, but I think I may have gotten lost in the shuffle of other
posts, and this prob is rather troubling.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Thursday, October 20, 2005 7:11 PM
To: 'Asterisk Users
You know, I was thinking I was going crazy... I've had that issue, and
another issue where whenever someone called one of our customers, the
customer would get 4 to 5 blank voicemails Still haven't tracked that
one down.
Sherwood
From: [EMAIL
Are you using SIP? If so, what version of Asterisk are you using?
In older CVS-HEAD versions, I _was_ getting sip deadlocks, but I haven't had
that problem in over a month...
Sherwood
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Andy Goss
-
- From: [EMAIL PROTECTED]
-[mailto:asterisk-users-
- [EMAIL PROTECTED] On Behalf Of Sherwood McGowan
- Sent: Friday, October 21, 2005 4:06 PM
- To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
- Subject: RE: [Asterisk-Users] asterisk locked up
-
- Are you using SIP? If so, what version
I've been poring
over the sample configs for the latest CVS-HEAD as well as the readmes from the
source's docs directory. I'm finding a lot of options that weren't previously
available, and would like to know if anyone's gone so far as to play with these
various new settings and document
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: Re: [Asterisk-Users] Any good docs for latest
-CVS-HEAD / Stable 1.2?
-
-Sherwood McGowan wrote:
- I've been poring over the sample configs for the latest CVS-HEAD as
- well as the readmes from the source's docs directory. I'm finding
Hey all,
I've a
rather annoying problem that I'd like to ask the users list about before passing
it on to the dev list.
I have my
asterisk system using SetCallerPres(prohibited_pass_screen) when a customer
requests an anonymous call. I have trustrpid set to yes, and sendrpid set to yes
Sorry, I should have specified, using a two day old
CVS-HEAD, and this is ALL in the SIP protocol
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowanSent: Thursday, October 20, 2005 7:05 PMTo:
'Asterisk Users Mailing List - Non-Commercial
Gents, this concerns a CVS-HEAD downloaded today.
I configured my system as I usually do, including using allowguest=yes
to attempt to correct the following problem, but to no avail. When any
call comes in from an external server I get this:
Oct 1715:36:43 NOTICE[4040]: chan_sip.c:10774
.
On 10/17/05, *Sherwood McGowan* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Gents, this concerns a CVS-HEAD downloaded today.
I configured my system as I usually do, including using
allowguest=yes
to attempt to correct the following problem, but to no avail. When any
and comments. I apologize in
advance if I missed something that was publicly available.
Sherwood McGowan
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that was publicly available.
-
- Sherwood McGowan
-
-
-
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I'm quite sorry mate, I didn't realize how you meant to connect it. I just
recently tried it because I decided to connect my router back into the local
network to find the system so that I could just have my * system take care
of all calls, etc...
I encountered the same problem. If you found any
line version of the ATA and/or router configs (wireless and
wired)
Thank you all in
advance for your thoughts and comments. I apologize in advance if I missed
something that was publicly available.
Sherwood McGowan
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I want to see if any
of my fellow Asterisk-Users list members have dealt with these guys. I'm a admin
for a VOIP provider, and have encountered a few PBX customers that want
consulting/support for the IAreaNet provided Asterisk pbxs. These guys are
selling AAH servers to the public, and are
You want something like this:
exten=_+1NXXNXX,1,SIPDtmfMode(inband)
exten=_+1NXXNXX,2,Wait(4)
exten=_+1NXXNXX,3,Playback(please-enter-your)
exten=_+1NXXNXX,4,Background(ha/mailbox)
exten=_+1NXXNXX,5,DigitTimeout,5
exten=_+1NXXNXX,6,ResponseTimeout,10
the information.
SKM
--Original Message-
-From: Stephen Bosch [mailto:[EMAIL PROTECTED]
-Sent: Thursday, September 29, 2005 12:18 AM
-To: Sherwood McGowan
-
-Hi, Sherwood:
-
-If you'll forgive me, I'd like to e-mail you directly with a
-few comments and questions.
-
-Sherwood McGowan
Let me ask you one major thing. Look at the CD filesystem on another
computer. Did you perchance burn the ISO as a file on the CD instead of
burning the ISO image to the CD?
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Anders Svensson
-Sent:
I have to barge in here...
Guys, the reason the audio sounds like hell is most probably for mulitple
reasons. First, what codec is the prompt encoded? Next, what codec is the
client using? If you're using, for example SIP clients with G711u, asterisk
has to re-encode (on the fly I might add) the
www.viatalk.com is
Asterisk friendly, and with purchase of a business plan (any of them) you can
get a toll free number. With the addition of a toll free number, we allow
multiple channels to be assigned to your account.
This solves the problem of multiple inbound/outbound
calls
Yes
- -Original Message-
- From: [EMAIL PROTECTED]
- [mailto:[EMAIL PROTECTED] On Behalf Of Waldo
- Rubinstein
- Sent: Sunday, September 25, 2005 9:06 PM
- To: Asterisk Users Mailing List - Non-Commercial Discussion
- Subject: [Asterisk-Users] Vonage-type service
-
- Is anyone offering
I know it's been
touched on before, but no answers have been found to the best of my knowledge.
I'm using a SIP only setup, with a sip provider giving PSTN and would like to
see if anyone has an idea for creating redial busy using ${DIALSTATUS} and
possibly MeetMe?
I figure something
like
Thank you, I do appreciate that many ATAs have redial on
busy, but I've been given the charge of figuring out how one would do it in
Asterisk.
Don't ask me why
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
EstepSent: Monday, September 26, 2005 10:15
Anyone else out there have some thoughts? The customer
wants to be able to control what can be redialed on busy, such as no
international. I'm having my doubts as to whether or not this can be done. My
idea seems like it would work, but after the customer hangs up, wouldn't the
context stop
FOP does this quite nicely
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
LaroffSent: Monday, September 26, 2005 1:57 PMTo:
Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] Extension
availabilty
I have a client that has an old Merlin system.
it should
just bridge the original caller back in.
On 9/26/05, Sherwood
McGowan [EMAIL PROTECTED]
wrote:
Anyone
else out there have some thoughts? The customer wants to be able to control
what can be redialed on busy, such as no international. I'm having my doubts
You may interested to know that a lot of connections will need inband when
speaking server to server. My system runs all users on RFC2833, except for
other asterisk servers. They run inband, because otherwise the DTMF wasn't
working.
Just my 0.02
--Original Message-
-From: [EMAIL
Actually, just point the line you want to use to a local ip
address (the asterisk server). I currently do this with my service. i.e. If your
Asterisk server is 192.168.15.200, just make the proxy for line1 that
address. It routes internally just fine.
Sherwood McGowan
From: [EMAIL
I have a need to use
cdr_custom and would like to know if anyone has gotten it to work with a mysql
cdr backend, and any examples if possible
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Try forcing the dtmf mode, such as
Exten=EXTEN,1,SipDTMFMode(inband)
That worked for me, but you'll need to only do it on the SIP calls, so route
accordingly
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Andrew Nowrot
-Sent: Thursday, September
It depends on the ATA, and our router, etc... Typically in the range between
1 and 2
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Adrien Laurent
-Sent: Monday, September 19, 2005 12:23 PM
-To: asterisk-users@lists.digium.com
-Subject:
Yes, because then the MACs specified would be getting the QoS, not just
certain ports. This is how I set up my customers when they have QoS
available.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Adrien Laurent
-Sent: Tuesday, September 20, 2005
the port?
-
-Sebastian
-
-
-
-On 9/20/05, Sherwood McGowan [EMAIL PROTECTED] wrote:
- Yes, because then the MACs specified would be getting the QoS, not
- just certain ports. This is how I set up my customers when
-they have
- QoS available.
-
- --Original Message-
- -From: [EMAIL PROTECTED
I'm looking to
upgrade my unit, and would like to not have to wait on our company's suppliers
to get back to me on it.
Thanks in advance
for any help
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I personally use Zabbix, and custom scripts that call
asterisk -rx or the manager API and return numeric digits. For instance, I use
"asterisk -rx 'sip show peers' | grep --text -i 'OK' | wc -l" to show me how
many users are connected at once, and test for no data return after 60 seconds
It's because mpg123 is being passed an option --mono. Looks to me (a cursory
guess) that your asterisk installation is trying to force mono sound, and
mpg123 doesn't like it.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Gurminder Arora
-Sent:
When using the
Comedian mail system, we've set up a remote access trunk to dump to the
VoicemailMain. Problem is, 90% of the time, the system won't recognize the key
tones from the PSTN. anyone else have this problem and end up solving it? We're
using Real Time setup, but the problem can't
Real quick guys, placing a ATA (such as a sipura SPA3K) somewhere similar to
a jack (like on the back of the nightstand where you placed the phone) would
be an easy solution to the problem of using SIP phones. That way customers
can use the normal office type phone or whatever standard phone you
Since I can't seem
to get anything figured out for the Comedian system, are there any other systems
out there that we can hook asterisk into?
Sherwood McGowan
ViaTalk
Level 2 Support
VOIP System Engineer
___
--Bandwidth and Colocation sponsored
seen your other posts. What exactly can't be figured out?
-
-Sherwood McGowan wrote:
-
- Since I can't seem to get anything figured out for the Comedian
- system, are there any other systems out there that we can hook
- asterisk into?
-
- Sherwood McGowan
- ViaTalk
- Level 2 Support
- VOIP
this functionality at a
-good price.
-
-Consider this post a blatant promo for comedian mail if you
-wish. I'm not ashamed to be using it.
-
-Steve Blair wrote:
-
-
- I haven't seen your other posts. What exactly can't be
-figured out?
-
- Sherwood McGowan wrote:
-
- Since I can't seem to get anything
?
- -
- - Sherwood McGowan wrote:
- -
- - Since I can't seem to get anything figured out for the
-Comedian
- - system, are there any other systems out there that we can hook
- - asterisk into?
-
-
-___
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Is WaitExten working
in current CVS-HEAD? I'm attempting to use it in some new dialplan code but not
getting anything
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That's almost always due to a NAT problem. Try using a stun server to solve
this problem (stun.gist.net for example)
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of Doug
-Sent: Thursday, September 08, 2005 11:38 AM
-To:
You'll wanna play with socket_read / socket_write Doing fsock_opens,
fgets(), etc... Is clunky
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Anton Krall
-Sent: Tuesday, September 06, 2005 1:06 PM
-To: 'Asterisk Users Mailing List -
There are unbundled Linksys devices, my company (ViaTalk)
uses them all the time
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karunakar
ChemuduguntaSent: Tuesday, September 06, 2005 3:16 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Wireless
I'm currently using
the Linksys PAP2, and since there's a shortage I'm looking for different
devices. I'm mainly looking at the Sipura SPA sets since they are the base of
the pap2. Anyone else have experience using them, and which
one?
Thanks
Sherwood
McGowan
one?
Thanks
Sherwood
McGowan
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http://lists.digium.com/mailman/listinfo/asterisk-users
Hey all,
I'm trying to create
redial on busy for my users, but haven't the foggiest on how to make asterisk
route depending on the status code returned over SIP (483, Busy Here?). . .
anyone know how to do this?
Thanks
Sherwood
McGowan
Has anyone else had
problems with users being able to press key tones during a voice prompt? I have
a few users complaining that some systems will not recognize key presses during
them.
using current
CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode.
Thanks
Sherwood
McGowan
: [Asterisk-Users] DTMF and breaking through
-voice prompts
-
-Sherwood McGowan wrote:
-
- Has anyone else had problems with users being able to press
-key tones
- during a voice prompt? I have a few users complaining that some
- systems will not recognize key presses during them.
-
-You are using
, and as such I'm unable to
bill...
Any
ideas?
Sherwood
McGowan
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To UNSUBSCRIBE
it so much I'm coding the
interface!
Sherwood McGowan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
LiSent: Friday, August 26, 2005 4:56 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] About
asterisk realtime
Hi,
I intend to use
Anyone have a good
tool(s) to use for simulating a bunch of calls? Benchmarking or stress
testing?
I only need SIP
protocol, and do appreciate any replies...I realize I could google it, but I am
looking for opinions as well.
Sherwood
McGowan
ARTCP (not yet released) will be doing exactly this, along with Zabbix for
monitoring (custom UserParameters will be included in ARTCP) for *REALTIME.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Darren Wiebe
-Sent: Friday, August 26, 2005 10:23
Anyone have a good
tool(s) to use for simulating a bunch of calls? Benchmarking or stress
testing?
I only need SIP
protocol, and do appreciate any replies...I realize I could google it, but I am
looking for opinions as well.
Sherwood
McGowan
Try broadbandreports.com and/or whichvoip.com...
I know of one company that can offer service, I work for 'em...ViaTalk
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of jennyw
-Sent: Thursday, August 25, 2005 3:17 AM
-To: Asterisk Users Mailing List
Sorry, sent with wrong accountread
below
From: Sherwood McGowan
[mailto:[EMAIL PROTECTED] Sent: Tuesday, August 23, 2005 8:34
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: SIP DEADLOCK
Anyone using a
CVS-HEAD pulled later than 8/13? We're runnign
since delete is a reserved word, what do you name a column in your
voicemail options table to allow setting of the delete option for realtime
voicemail? Anyone?
Sherwood McGowan
ViaTalk
Level 2 Support
VOIP System Engineer
___
Asterisk-Users
is kinda why the -x option as
added anyway...
Anyone?
Sherwood
McGowan
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]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Eric Wieling aka ManxPower
-Sent: Monday, August 22, 2005 3:15 PM
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: Re: [Asterisk-Users] asterisk -rx (or remote
-connections in general)
-
-Sherwood McGowan wrote:
- I haven't been able
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: Re: [Asterisk-Users] asterisk -rx (or remote
-connections in general)
-
-Sherwood McGowan wrote:
- Now, there actually is actually documented problems with too many
- remote connections to the manager (CLI). . . I'm asking
Of
-Sherwood McGowan
-Sent: Monday, August 22, 2005 2:14 PM
-To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
-Subject: RE: [Asterisk-Users] asterisk -rx (or remote connections in
-general)
-
-Maybe I am, I don't doubt it.
-
-But why does asterisk deadlock then when about 5 or 6 scripts
Thanks for the hint. I'm looking into using a manager proxy and just
recoding to use the manager interface/proxy for my scripts.
Thanks for your help,
Sherwood McGowan
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Eric Wieling aka ManxPower
Is this a CVS-HEAD that was released AFTER 8/13? I ask because we're using
that release and it's still deadlocking.
Thanks,
Sherwood McGowan
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Kevin P. Fleming
-Sent: Saturday, August 20, 2005 3:25 PM
.
Sherwood McGowan
ViaTalk
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Justin Richards
-Sent: Saturday, August 20, 2005 8:29 AM
-To: asterisk-users@lists.digium.com
-Subject: [Asterisk-Users] ViaTalk Down?
-
-Is anyone else with ViaTalk experiencing
I've restarted our switch via restart command from the CLI.
Anyone have a quickie answer as to why asterisk would suddenly just stop
responding? I was able to issue the restart command but I couldn't do sip
show peer num and couldn't show channels, etc This is very
disconcerting
We've
Last note on this, I figured out it was due a freeze in registrations that
we've been having an issue with on asterisk. I'm writing a custom monitoring
script using sipsak for testing registrations, which would SMS the
engineering dept when registrations stop working.
Cheers,
Sherwood McGowan
Has anyone been able
to fix the issue of too many Asterisk remote connections on the CLI? I'm using
monitoring scripts that use asterisk -rx 'command' and some of them hang (that
doesn't make sense either). The hangs cause open connections to build up (still
not that large) until the
This is not an answer but rather an addition to the question. We're using a
large scale VOIP only asterisk system that has PAP2 enduser units using
inband as their DTMF mode. sip.conf is set for using inband as well, and we
pass PSTN calls through a provider.
Here's the problem, when our users
Of
-Sherwood McGowan
-Sent: Tuesday, August 16, 2005 8:43 AM
-To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
-Subject: DTMF, Asterisk, External PSTN gateway,and PAP2 (was: RE:
-[Asterisk-Users] Issue with DTMF Tones - CodecIssues)
-
-This is not an answer but rather an addition
-gateway, and PAP2
-
-Sherwood McGowan wrote:
- I'll pass that on to my lead engineer, he was under the assumption
- that
- rfc2833 was too unreliable. I personally don't know, but will look
- further into the matter.
-
-You need a new engineer. OOB DTMF like RFC2833 is more
-reliable than inband
Great job! PhoneCall is very similar to the interface that I'm writing now,
with many of the same features! I love it!
Sherwood McGowan
--Original Message-
-Dustin Wildes wrote:
- Hello All!
-
- Just a notice that our PHP/Smarty-based GPL version of PhoneCALL
- version
- 2.6.1 has
I could be wrong butsip / iax prune realtime user
[user] ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
EstepSent: Tuesday, August 16, 2005 5:10 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] realtime
As far as I remember, you can't really do that (because the telco isn't
switching the call), what you'll want to do is have a hunt group set up
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Innocent Evil
-Sent: Monday, August 15, 2005 2:17 PM
.
Sherwood McGowan
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Innocent Evil
-Sent: Monday, August 15, 2005 2:29 PM
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: Re: [Asterisk-Users] Switch between FXS ports
-
-Sorry
for donations before at least some cursory info has been
released to show that the project is at least happening ;)
Cheers all, and I hope to see interest in getting this going.
Sherwood McGowan
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Side note, I've jumped to a different name, as ARTCP defines more the
control program portion than an entire distro.
ARTP -Now- AstCD (Asterisk Complete Distribution)
Obviously the name would change to something a little more memorable once
the project is in a release phase
Sherwood
on something similar atm, schemas written,
-etc) - but at the end of the day mysql still does not cut it inmo.
-
-No offence to mysql developers, etc.
-Cheers
-Chris
-
-- Original Message -
-From: Sherwood McGowan [EMAIL PROTECTED]
-To: 'Asterisk Users Mailing List - Non-Commercial Discussion
, September 23, 2004 11:19 PM
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: Re: ARTCP Project (was RE: [Asterisk-Users] Features
-you'd like toseeina GUI?)
-
-Hi,
-
-Sherwood McGowan wrote:
- I personally prefer MySQL-MAX. I curently run *RT in a large
- production environment
out of my head and into a working system, and possibly
helping other people who might need it at the same time.
Cheers,
Sherwood McGowan
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-snacktime
-Sent: Friday, August 05, 2005 1:04 PM
-To: Asterisk
He's right. My company uses Linksys PAP2s, and yes they do have stutter tone
and short ring options in the box.
We also use MWI, and have had no issues with it at all
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Matthew Boehm
-Sent: Friday,
seen many RT
versions, which I prefer overall.
I'm definitely starting up a site for information and the option to
contribute to the development of it. I'll post to this list when I get it
out there.
Sherwood McGowan
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED
.
Thanks again,
Sherwood McGowan
ARTCP Developer
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I experience this problem as well, a 512MB machine drops down to less than
10MB. I'm running *RealTime however.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Greg Boehnlein
-Sent: Thursday, August 04, 2005 10:35 AM
-To: Asterisk Developers
Messaging
-Business in a Box 4)Asterisk Hosted Telephone and PBX
-business in a Box
- Etc etc.
-
-My .02 cents on this.
-
-Seshu Kanuri
-
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Sherwood McGowan
-Sent: Thursday, August 04, 2005 9:02 AM
-To: 'Matt
-Business in a Box 4)Asterisk Hosted Telephone and PBX
-business in a Box
- Etc etc.
-
-My .02 cents on this.
-
-Seshu Kanuri
-
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Sherwood McGowan
-Sent: Thursday, August 04, 2005 9:02 AM
-To: 'Matt Florell
, not upon boot
every time, as the Sipura-SPA might (although Linksys's ATAs have sipura-spa
backends, AFAIK, let me know if I'm wrong).
I do agree though, with the point of multiple traversals of segements and
firewalls, and even the point of US to Japan being a possible problem.
My $0.02
Sherwood
We use Asterisk RealTime for our VOIP Telephone service all over the
country. It is definitely more of a switch. We use an outside provider to do
the transition of VOIP to POTS though, as it would be astronomically
expensive to provide US/CAN free longdistance otherwise, even with a switch
for
Does anyone know if
AbsoluteTimeout is working completely? As far as I can see on my systems, I'm
still getting occasional hung SIP channels, even though there should be nothing
over my setting...
cheers
Sherwood
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the features for each
would be different.
Let me know, as I'd like to try to get as much as possible
right the first time.
Cheers,
Sherwood McGowan
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I can say that MWI is working with RT in my particular system, and I'm
running several hundred users.
Sherwood
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Polycom User
-Sent: Wednesday, August 03, 2005 8:07 PM
-To:
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