On Wed, 21 Apr 2004, Ben Merrills waxed:
> Hi,
>
> I have a couple of questions about MeetMe and call queues. I'm still
> pretty new to Asterisk, but already having to write a Service Center
> call manager for it (which I might add, our director has agreed to make
> open source!).
That's great
You should checkout [EMAIL PROTECTED], that's what we have installed here. It gives you web access to voicemail as well as a web configuration tool (Asterisk Management Portal). For the POTS line, you will need an FXO card. We are a Digium reseller so I can get you what you need as far as hardware.
[EMAIL PROTECTED] wrote:
[SNIP]
line, you will need an FXO card. We are a _Digium reseller_ so I can get
[SNIP]
other questions, I just started using Asterisk about _2 weeks ago_. I know
[SNIP]
How do you end up being a Digium reseller after using Asterisk for two
weeks? Do you plan to prov
I've got my own Asterisk tech. He's been with the company for about 2 years, finally convinced me to switch to Asterisk.Matt Riddell <[EMAIL PROTECTED]> wrote:
[EMAIL PROTECTED] wrote:[SNIP]> line, you will need an FXO card. We are a _Digium reseller_ so I can get [SNIP]> other questions, I just st
On 6/5/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> You should checkout [EMAIL PROTECTED],
Thanks for the link - unfortunately [EMAIL PROTECTED] won't quite do the trick
for
me here as it is a complete OS replacement from what I can tell... I
can't do that. I have too much time and money i
L PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey
> Sent: Monday, 6 June 2005 10:09 AM
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] A Few Questions
>
> On 6/5/05, [EMAIL PROTECTED] <[EMAIL P
PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey
Sent: Monday, 6 June 2005 10:09 AM
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] A Few Questions
On 6/5/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]>
wrote:
You should
t; >
> >
> >>-Original Message-
> >>From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >>[EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey
> >>Sent: Monday, 6 June 2005 10:09 AM
> >>Cc: Asterisk Users Mailing List - Non-Commercia
On Mon, Jun 06, 2005 at 03:38:52PM -0400, Matt wrote:
> Then download the tar ball! It will install on any redhat (and maybe
> other) based systems as it compiles from source.
Great. Get the latest RedHat for FreeBSD from http://redhat.com/bsd/ .
Naturally the linux zaptel code will compile clean
In trying to follow Marks advice and be nice to newcomers, I'll just put
URLS below.
On Tue, 2004-08-03 at 06:10, Mark wrote:
> We have several C/T servers with PRI lines that are under utilised, in
> the following configuration
>
> eISDN -> PRI -> C/T Server 1
>
> eISDN -> PRI -> C/T Server 2
>
Overhead paging is totally possible, there are several articles available on
how to do it. But you cannot have multiple zones today unless you use a sip
device that has autoanswer.
Easiet way to remove that message is to replace the file with one that only
has a split second of silence.
-Or
That article is about shared call appearance. I have this working using
Grandstream GXP-2000's. It's a great new feature.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Friday, December 09, 2005 8:21 AM
To: asterisk-users@lists.
On 12/9/05, Stas Khromoy <[EMAIL PROTECTED]> wrote:
> we are beginning to test asterisk for our office
> one of the features of the current phone system that is very heavily
> used is overhead paging
Overhead paging can be done with asteirsk in anyway you want, you can
even do mutilple zones, all
On 12/9/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> Overhead paging is totally possible, there are several articles available on
> how to do it. But you cannot have multiple zones today unless you use a sip
> device that has autoanswer.
>
Why can mutilple zones not be done?
hehe
I just installed * with a T1 span to and Adit600 with 2fxs and 1fxo
The 8 fxo ports were for zone pageing
works great
should work with any fxo device and an existing page system
On Dec 9, 2005, at 11:34 AM, C F wrote:
Overhead paging is totally possible, there are several articles
avai
On 12/9/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> Overhead paging is totally possible, there are several articles
> available on how to do it. But you cannot have multiple zones today
> unless you use a sip device that has autoanswer.
>
>Why can mutilple zones not be done??
List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] a few questions
On 12/9/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> Overhead paging is totally possible, there are several articles
> available on how to do it. But you cannot have multiple zones today
> unless
my apologies if anything
but as i said i am not that knowledgeable
and most probably misunderstood the post.
as it looks from your reply i have
if you don't mind letting me know what i got wrong
i would greatly appreciate it.
On 12/9/05, Stas Khromoy <[EMAIL PROTECTED]> wrote:
we are beginn
On 12/9/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> On 12/9/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> > Overhead paging is totally possible, there are several articles
> > available on how to do it. But you cannot have multiple zones today
> > unless you use a sip device that has autoansw
Hey - best trick is not to take any of it personally.
We all fell off our bikes while learning to ride!
(old quote, but still valid)
PaulH
> Stas Khromoy <[EMAIL PROTECTED]> wrote:
>
> my apologies if anything
> but as i said i am not that knowledgeable
> and most probably misunderstood the po
Simon -
There are certainly some things that are lacking with Asterisk's
SIP implementation, some of which you may have tripped over. Of
course, Asterisk's SIP stack is "pragmatic" and has been developed as
people discover needs for particular functions. However, there are
perhaps some solu
> Is this correct?
I see the 100 Trying on REGISTER frequently, but if it's not valid, we can
take it out. It serves no really effective purpose.
> 2. 10.3 Processing REGISTER requests. The 5th paragraph states "that the
> registrar has to know the set of domain(s) for which it maintains
> bindi
> >1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue
> >a provisional response to non-INVITE requests.
> >
> >>From my message yesterday * appears to be sending a SIP/2.0 100 Trying to
> >X-Lite's REGISTER request before sending the SIP/2.0 200 OK message.
> >
> >Is this c
[EMAIL PROTECTED] (John Todd) writes:
>There are certainly some things that are lacking with Asterisk's
> SIP implementation, some of which you may have tripped over. Of
> course, Asterisk's SIP stack is "pragmatic" and has been developed as
> people discover needs for particular functions.
[EMAIL PROTECTED] (Mark Spencer) writes:
> > Is this correct?
>
> I see the 100 Trying on REGISTER frequently, but if it's not valid, we can
> take it out. It serves no really effective purpose.
I think that it's only on REGISTER messaegs that it shouldn't be
used. Perhaps previous RFCs didn't
> > >3. I have another SIP account (sip:[EMAIL PROTECTED]) which I would like
> > >to use within asterisk both for dialing out and for receiving calls.
> > >
> > >I see that sip.conf has a line
> > >
> > >register => [EMAIL PROTECTED]/1234
> > >
> > >where 1234 is the local asterisk extension. Fro
Hi Mark,
[EMAIL PROTECTED] (Mark Spencer) writes:
> > register => [EMAIL PROTECTED]:[EMAIL PROTECTED]:port/localpart is
> > acceptable I can look at modifying the existing source to accept this
> > format and using it appropriately. That shouldn't be too hard (I
> > think).
>
> Actually I just
Hi
There is a device called a "parlay" made by a crowd called
voxtream which will route the ISDN calls based on the DID
and/or the callerid, before the call is answered.
It would be nice if this feature could be done in Asterisk
as well, but at this point in time, it first answers the
call.
reg
On Tue, 3 Aug 2004 [EMAIL PROTECTED] wrote:
> There is a device called a "parlay" made by a crowd called
> voxtream which will route the ISDN calls based on the DID
> and/or the callerid, before the call is answered.
>
> It would be nice if this feature could be done in Asterisk
> as well, but a
On Wed, 4 Aug 2004 13:34:02 +0200 (CEST)
Peter Svensson <[EMAIL PROTECTED]> wrote:
> On Tue, 3 Aug 2004 [EMAIL PROTECTED] wrote:
>
> > There is a device called a "parlay" made by a crowd
> called
> > voxtream which will route the ISDN calls based on the
> DID
> > and/or the callerid, before the
On Mon, 2005-09-05 at 01:31 -0400, Kurth Bemis wrote:
> I am attempting to assemble a proposal for a client of mine that is
> looking to replace their phone system. I think it's a good first
> installation with 4 POTS incoming and 15 extensions, with an overhead
> paging system. I also think tha
Adam,
Thanks for your help.
Does anyone know or is anyone an * guru in the New Hampshire/Vermont area?
how about this example.
User1 sits at his desk, a call comes in.(doesn’t matter how the call
gets to his phone, DID or exten) he needs to go into the warehouse to
look at something. He pla
Hi Kurth,
I'm in NJ. I'd be happy to help you out either on the phone or in person.
Gimme a call 973 828 1625
Mark
Kurth Bemis wrote:
Adam,
Thanks for your help.
Does anyone know or is anyone an * guru in the New Hampshire/Vermont area?
how about this example.
User1 sits at his desk, a ca
>>From: "Mark Scholten" m...@streamservice.nl
>>Hello,
>>I have a few questions regarding Asterisk 1.8.0. If you can answer a
>>question, please do so.
>>Is Asterisk 1.8.0 stable enough for production environments?
It appars to be so far we are testin
> Good luck as with any new version there may be some bugs so if you bump up
against ones report them so they can be fixed.
> Also don't just drop it into production with out testing it on a box for a
bit. 1.8 has a lot of changes. Most appear to be for the better.
The only important difference I
Dmitriy Serov wrote:
At the moment I plan to migrate from asterisk 13.7 to 13.8.
Because of relatively frequent updates I am building asterisk from a
directory that is updated via git switch to the desired branch.
1. Will be updated pjproject patches with "git pull"?
Yes.
2. Will update hims
On Sun, Jun 05, 2005 at 11:06:48PM -0400, C. Hatton Humphrey wrote:
> I have Asterisk running on a FreeBSD machine that is also my
> router/firewall and MySQL server.
Asterisk is a CPU-intesive program. It will probably work fine with a
router/firewall, but with another potentially-CPU-intensiv
Warren wrote:
Am I best off using Hylafax?
Yes.
Lee.
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Hi Warren -
Questions:
(1) Good 2xT1 card with hard echo cancellation?
I'm not sure if it has onboard echo can, but Sangoma has a two port
model. I've never used them (I have a TE410P), but I've always read
very positive things about them, especially on the quality of the echo
can they use.
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