> Daniel Johnson wrote:
>
> Jeff LaCoursiere wrote:
> > I think you are looking to use a "campon" feature. Try this:
> >
> > http://www.voip-info.org/wiki/view/Asterisk+tips+campon
> >
> > j
> >
>
> Hi Jeff,
>
> Yes I have seen this feature. Its a half implementation of what we
> requi
> Daniel Johnson wrote:
>
> pdha...@optusnet.com.au wrote:
> > Funnily enough, most people install phones with BLF lamps, on install
> something like hudlite/FOP/etc so you know if the person is on the phone
> before you call them..
> >
> > PaulH
>
> Hi Paul,
>
> Yes I have seen these too
> Daniel Johnson wrote:
>
pdha...@optusnet.com.au wrote:
> Funnily enough, most people install phones with BLF lamps, on install
> something like
hudlite/FOP/etc so you know if the person is on the phone before you call them..
>
> PaulH
Hi Paul,
Yes I have seen these tools. However it is a
Funnily enough, most people install phones with BLF lamps, on install something
like hudlite/FOP/etc so you know if the person is on the phone before you call
them..
PaulH
> Daniel Johnson wrote:
>
> Hi,
>
> I am trying to implement the callback feature of our old phone system.
> This fea
If machine 1 has a zaptel card in it...and not running asterisk...and another
machine with no card, but running Asterisk...
Me confused!
Move the card? Move the Asterisk?
PaulH
> Brian J. Murrell wrote:
>
> Slightly OT, but I'm wondering if anyone here has come across a "soft
> ATA". T
Sure - read up on IAX for a few good points.
PaulH
> rupak shrestha <[EMAIL PROTECTED]> wrote:
>
>
> Hi all, i have a basic question on asterisk.The below is my scenerao.
> I have my sales offices around the globe.Theyare all connected with
> Speed Internet connection.I don't mind installing
We are looking for a site running Queumetrics in Sydney, Australia.
We have been contacted by a company in Sydney, as a few staff members of a
company that are currently running Queuemetrics would like to see a fully
running installation for training and decision making purposes. Their trial
l
The new grandstream video phone has rca-style audio jacks
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
- Original Message -
From: "Julien Goodwin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, May 20, 2006 9:30 P
I thought the price of E1 in Australia was quite reasonable, at least
compared to analog.
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
- Original Message -
From: "Craig Guy" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday
In our case, it was cpuspeed (a daemon) interfering with the zaptel drivers.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Rich Adamson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, April 27, 2006 10:14 PM
Subjec
Each call has a unique callid - I used that for a dialplan a short while
ago, to do a very similar job to what you are doing...
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Shaun" <[EMAIL PROTECTED]>
To:
Sent: Sunday, April 16, 2006 5:06 PM
Subject: [Asterisk-User
We had an issue at an install of [EMAIL PROTECTED] - where if you use the
external extensions the machine is unable to start Asterisk after a reboot.
Which in the end begged a question - it was nice have customers who could edit
their box, but was it worth it for the angry calls when their PABX
We have written something, so the answer is yes.
regards,
Paul Hales
Technical Manager
AsteriskIT
> Joseph <[EMAIL PROTECTED]> wrote:
>
> Is there a way somehow to implement Asterisk with Credit Card Processing
> (IVR system)?
>
> --
> #Joseph
> __
I meant dialled extension, not originating
extension.
like :
exten =>
_37X,1,Dial(IAX2/FAX/${EXTEN})
exten =>
_38X,1,Dial(IAX2/NOTFAX/${EXTEN})
Paul HalesTechnical
ManagerAsteriskIT
- Original Message -
From:
Michael
Strelnikov
To: Asterisk Users Mailing List
What about different extensions using different connections?
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Brian Capouch" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, April 10, 2006 4:26 AM
Subject: Re: [Asterisk-U
WaitExten perhaps?
Paul Hales
Technical Manager
AsteriskIT
> Dennis <[EMAIL PROTECTED]> wrote:
>
Hi,
Let me explain a little about our system here first.
We have a Digium TE110P card hooked up to an isdn 30 line. (Australia -
EuroISDN) This works fine.
We also have a Digium TDM400P wit
It means that you are loading the digium card up with incorrect values.
I had it happen to me recently.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Jon Farmer" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, March 28, 2006 9:31 PM
Subject: [Asterisk-Users] Problems with
You can get an extension module that adds another 42 buttons.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Daniel Hazelbaker" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, March 28, 2006 8:22 AM
Subject: Re: [Aste
Yes - set up about 10 of them at a business last year.
Monitoring is fine - picking up calls is a bit iffy at the best of times.
(that is, picking up a ringing call by pushing the extension button. *8
works fine)
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Darrel
Using a Snom phone, you can monitor a lot more extensions, so I figure it's
got to be a Polycom issue.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Daniel Hazelbaker" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday,
Understanding..is not required. ;)
PaulH
- Original Message -
From: "James Harper" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, March 27, 2006 12:23 PM
Subject: RE: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
> > Not GS
Now that's an interesting comment - most people think the speakerphone on
the Polycom is quite good.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Denis Galvão - iSolve" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Cc: <[EMAIL P
I had a look at the snom website - and the manual for the expansion module
read that only one module can be attached 'currently'.
So maybe this has changed. Any ideas?
Personally, I like snom phones a lot. I used a snom 200 at my desk at a
previous job for almost 2 years.
Paul Hales
Technical Man
I installed 2 Snom360's a few months ago, and 'at the time' only 1 expansion
module could be added.
(also the fact that the modules draw so much current that it got the POE
switch upset!)
Have you tested a snom360? I should have one in the lab soon enough.
Paul Hales
Technical Manager
Asteris
I have a bad feeling that getting a phone with 160 lights is not going to
happen anytime soon.
>From memory, the snom360 is limited to way less than that.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Stuart Elvish - Dallas Delta Corporation Pty Ltd"
<[EMAIL PROTEC
And the fact that rebooting a phone is a fairly rare occurence.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Avi Miller" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion"
Cc: <[EMAIL PROTECTED]>; "Kevin P. Flemi
I think the main issue for James and myself is that we can't buy anything in
Australia.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "AR Tarzi" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, March 27, 2006 10:21 AM
S
What about using system(echo) to push stuff into a text file, or the mysql
plugin to push stuff over to a database?
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Clint Tevlin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: S
If you find anything out, I would like to know.
I have tried to find a gsm/wifi phone in the past (in melbourne) and failed.
later,
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "James Harper" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Di
If you can find yourself a local Asterisk consultant, they should be able to
let you see some phones and maybe even try them out.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Daniel Hazelbaker" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial D
What sort of call path are you trying to get working?
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, March 26, 2006 10:18 AM
Subject: [Asterisk-Users
We ran a system at one site with 2 TDM400's in it to hook up to 8 analog mobile
phone gateways.
Asterisk was much more reliable than the analog phone gateways, but we still
rebooted it once a week.
Running on a dual athlon 1800 we picked up very cheaply.
regards,
Paul Hales
Technical Manager
You will most likely need an E1 crossover
cable.
With regards to the dialplan programming, E1
connections (internal and external) are much the same - so dial(ZAP/G1) and so
on will work fine.
regards,
Paul HalesTechnical
ManagerAsteriskIT
- Original Message -
From:
r
Audacity?
PaulH
- Original Message -
From: "Douglas Garstang" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 22, 2006 8:25 AM
Subject: RE: [Asterisk-Users] Native MOH - Convert mp3 to ulaw
> Good grief. Considering all the libr
User - sends you calls
Peer - you send calls
Friend - Both ways
later,
Paul HalesTechnical
ManagerAsteriskIT
- Original Message -
From:
Douglas
Garstang
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, March 22, 2006 5:07
AM
Su
You will have to break up the card into 4 Zap
channels, and them use some clever dialplan work to make sure people are using
the right channels.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From:
Faisal
Inam
To: asterisk-users@lists.digium.com
S
I was also thinking a list for newbies...
PaulH
- Original Message -
From: "Robert La Ferla" <[EMAIL PROTECTED]>
To:
Sent: Saturday, March 18, 2006 2:33 PM
Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic
> The volume/traffic on this list has been getting pretty heavy.
But it looks like a step in the right direction.
PaulH
- Original Message -
From: "James Harper" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, March 18, 2006 4:49 PM
Subject: RE: [Asterisk-Users] gsm picocells
> > I believe the OP wa
Any idea where I can get some of these units in Melbourne?
Paul Hales
AsteriskIT
> Faxing received by SpanDSP seems to work fine with these units.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Phelan
Sent: Tuesday, 14 March 2006 9:00 AM
To: 'A
Where are you located?
We are in Melbourne, Australia.
regards
PaulH
- Original Message -
From: "Naren Koka" <[EMAIL PROTECTED]>
To:
Sent: Monday, March 13, 2006 11:25 PM
Subject: [Asterisk-Users] Need help implementing call center features
ofAsterisk
> I am looking for help in impl
Noted - I may need to grab one for an install coming up.
regards,
PaulH
- Original Message -
From: "Avi Miller" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, March 09, 2006 10:11 PM
Subject: Re: [Asterisk-Users] MultiBRI in Australia
Shove a WaitExten at the end of the menu.
PaulH
- Original Message -
From: "Robert P. McKenzie" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, March 10, 2006 8:21 AM
Subject: Re: [Asterisk-Users] IVR woes
> Sean,
>
> Thanks I've made th
WaitExten is also a great item to put at the bottom of your menus.
PaulH
- Original Message -
From: "Sean Cook" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, March 10, 2006 8:13 AM
Subject: Re: [Asterisk-Users] IVR woes
> -BEGIN PG
Totally correct - according to me at least.
PaulH
- Original Message -
From: "Ken D'Ambrosio" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Cc: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion"
Sent: Tuesday, March 07, 20
Can you provide a photo of this?
I am interested in seeing it!
PaulH
- Original Message -
From: "Douglas Garstang" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, March 07, 2006 2:13 AM
Subject: RE: [Asterisk-Users] Polycom 501 power ove
I have installed several hundred polycom's, and I have never seen a 500/501
with a power jack.
All with the inline cable, as you mention.
Of course, if someone can provide photo evidence I will stand corrected.
PaulH
- Original Message -
From: "The VoIP Connection" <[EMAIL PROTECTED]>
T
Just trying to think - are you using the standard E1 setup from ATP?
I have found that the settings on their website work pretty well.
Also - have you tried to put an answer in your dialplan? That might keep the
dialplan open..
later,
PaulH
- Original Message -
From: "Paul C" <[E
Funnily - I have set up 2 or 3 pri's over the last few weeks on 1.2x and
haven't had any issues.
(and one of those is a high load situation - passthru at an outbound call
centre)
PaulH
Melbourne
- Original Message -
From: "James Sturges" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing
I actually got it all working - but it's great to see where we did the same
thing, and where we differ.
I ended up using the 'pop' perl command - inside a loop to go back one item
at a time through my list
PaulH
- Original Message -
From: "Michael Collins" <[EMAIL PROTECTED]>
To: "A
Thanks to everyone who helped me with this!
I will post my code next week (when I am back at the workplace where I did
this)
later,
PaulH
- Original Message -
From: "Paul Hales" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 01
Ummm - from memory the sequence is TRANSFER
-> NUMBER -> SEND -> chat to other person -> TRANSFER.
PaulH
- Original Message -
From:
MBIT Technologies
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Thursday, March 02, 2006 11:02
PM
Subject
I need to read through the numbers in reverse order, so I can decide which
messages to play to people.
I was using a variable to mark how many messages they had read, and each
time read a number further back in the list.
PaulH
- Original Message -
From: "Michael Collins" <[EMAIL PROTECT
That's getting pretty close - thanks for that.
I just couldn't find any decent info on the web about working with AGI.
regards,
PaulH
- Original Message -
From: "Michael Collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 0
Numbers from a text file - just need to read them back in one at a time.
PaulH
- Original Message -
From: "Michael Collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 01, 2006 5:14 AM
Subject: RE: [Asterisk-Users] Re: Asteris
I was going to see if I can execute a bash script as an AGI - just looking
around the internet for examples at the moment.
Anybody got an example spare?
I'm just a bit stuck on how to start this, but I am quite comfortable writing
asterisk dialplan stuff and bash scripts
later,
PaulH
> s
It's the 'o' extension in your context that hits the voicemail.
(thats a lower case o not a zero)
PaulH
- Original Message -
From: "Paul Tinsley" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 22, 2006 3:19 AM
Subject: [Asteri
It's funny, but I found it more challening to buy a second hand car than to
buy phones.
PaulH
- Original Message -
From: "mustardman29" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, February 22, 2006 4:57 AM
Subject: [Asterisk-Users] What business IP phone to use
>
>
> I have been struggli
Jenn bought hers at Borders(jenn @ ains from this list)
PaulH
- Original Message -
From: "Bob McDowell" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 22, 2006 5:32 AM
Subject: RE: [Asterisk-Users] Download "Asterisk: The
This sounds more like a dialplan issue - and what has got to do with
anything?
PaulH
- Original Message -
From: "Alejandro Vargas" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, February 21, 2006 8:16 PM
Subject: [Asterisk-Users] i
Where are you located?
Paul Hales
Melbourne, Australia
- Original Message -
From: "Sam Tam" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, February 21, 2006 8:52 PM
Subject: [Asterisk-Users] $ for an hr of asterisk support
> Hello
>
Depending on the setup, you might actually need 8 x
FXO.
PaulH
- Original Message -
From:
Marc
Archer
To: asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2006 10:24
AM
Subject: [Asterisk-Users] Multiple
TDM400P's in a single machine
Hi
But using the native transfer on the phone causes the system to think the
agent is still on the call
Is this still correct? It was last time I tested it, 6 odd months ago.
PaulH
- Original Message -
From: "Eric "ManxPower" Wieling" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing Lis
Where are you located? That makes a big difference!
PaulH
Melbourne, Australia
- Original Message -
From: "CyberSource" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, February 21, 2006 7:37 AM
Subject: [Asterisk-Users] good voip
> Can anyone recommend a good voip provider? Thanks
> __
- Original Message -
From: "Tomislav Parčina" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, February 20, 2006 10:46 PM
Subject: [Asterisk-Users] Re: Call centre - * hang's up
> In article <[EMAIL PROTECTED]>,
[EMAIL PROTECTED] says...
>
On an Asterisk server - yes.
[EMAIL PROTECTED] -
not me.
PaulH
- Original Message -
From:
Dean
Collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, February 20, 2006 1:17
AM
Subject: [Asterisk-Users] Wildfire
messsaging s
To translate between g729 and g711 you need to buy some licences.
PaulH
- Original Message -
From: "Lisa Wolf" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, February 16, 2006 11:38 AM
Subject: Re: [Asterisk-Users] Connecting two phones
The genzaptelconf doesn't work with E1/T1 cards in my experience.
You will have to configure it by hand.
PsulH
- Original Message -
From: "nik600" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, February 11, 2006 11:09 PM
Subject: [Aster
There are quite a few Asterisk systems running in Australia, so you should
be fine
PaulH
Melbourne
- Original Message -
From: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, February 10, 2006 3:31 AM
Subject: [Aste
> "Alexander Lopez" <[EMAIL PROTECTED]> wrote:
>
> >>
> You can do all that you want and MORE with 'hand-knitted' configuraton
> files.
> <<
>
> That's what I figured; I was really wondering whether AMP would
> specifically *prevent* that kind of configuration.
>
> I'm presuming that a hand-knitte
> >>
> >> Thanks for the answers. I really appreciate that. It may be better
> for
> >> me to talk to local Telco for further price negotiation.
> >>
> > Going through a VOIP termination service is also good for testing.
> >
> When going thru a VoIP termination service, do I also need to have a IP
>Hi,
>
>I m still not quite understand why I need E1/T1 PRI line with VoIP calls
>to normal phone line. I thought I can send calls out thru a normal home
>telphone line. If this is the case, I will just need to pay each VoIP call
>to phone line at 20 - 30 cents / call.
Then you will need to buy an
> Hi,
>
> We have 10 people on our network and each person will have a SIP phone
> connected to our Asterisk server. All phones, Asterisk, other servers and
> users workstations will be using the same network. The question is: would
> I need a QOS device to give SIP traffic a chance? Our interna
> >There are lots of Asterisk users here in Australia...and it's not
illegal.
> >
> >You will probably have to discuss charging with your Telco.
> >
> >PaulH
> >
> >_
> >
> >
>
> Thanks for the answers. I really appreciate that. It may be better for
> me to talk to local Telco for further price neg
> >
> > How much your telco is going to charge you for the PSTN calls depends
> > on your arrangement with the telco... Usually, with proper volume
> > interconnects (say you order a PRI line), these calls are charged per
> > second.
> >
> Do I really need PRI T1 line when I initially setup VoIP
> Thanks for the answer. Is this PSTN gateway is something for a VoIP
> company to setup in order to connect their VoIP calls to the Telco's
> PSTN then to the end phone? I don't think Australia treat this as
> illegal. But I m not sure how much the Telco will charge from IP PAX (or
> PSTN) gat
Shouldn't all sip users have different usernames?
(or am I missing some vital detail here?)
PaulH
- Original Message -
From: "Miguel" <[EMAIL PROTECTED]>
To: "Asterisk User List"
Sent: Friday, February 03, 2006 3:21 AM
Subject: [Asterisk-Users] limit sip sessions
> hi, is there a way
h - using hylafax or asterisk?
PaulH
- Original Message -
From: "James Harper" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, February 02, 2006 10:13 AM
Subject: [Asterisk-Users] fax possibilities
> I am trying to set up a linux b
Do you mean 1-800 number?
I don't really know the answer - I will have to ask
next time I visit.
PaulH
- Original Message -
From:
Joe Pukepail
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, February 02, 2006 7:47
AM
Subject:
I think they have a 1-800 number so you might be right.
But the important question is still - will Asterisk support this?
PaulH
> Alexander Lopez <[EMAIL PROTECTED]> wrote:
>
They are using ANI instead of CallerID. If they have an 800 number thya
have the right to know who is calling them beca
I think the customer is more interested in whether Asterisk can do what their
current system does, rather than discuss the legalites of this.
PaulH
> Chris Bagnall <[EMAIL PROTECTED]> wrote:
>
> > The issue is that their dialogic solution can read callerid
> > from incoming calls, even if the
If callerid is received, it will be displayed on the sip phones.
My guess would be that it's not coming in on the analog line in the first place.
PaulH
> Scott Geist <[EMAIL PROTECTED]> wrote:
>
> How do you retreive the caller id on incoming analog lines and display
> the
> id on the sip pho
Is ztdummy loaded properly?
I had a similar problem with a system recently.
PaulH
- Original Message -
From: "Hadar Pedhazur" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, February 02, 2006 8:16 AM
Subject: [Asterisk-Users] No Audio o
You are missing the G
exten => 190,1,Dial(ZAP/g0/800111)
Assumes that you have set up your zap card as group 0.
(zap/g1 is probably more realistic)
later,
PaulH
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Thursday, February 02, 2006 7:11 AM
Subject: [Asterisk-Use
I have been discussing an asterisk solution with a
company that has a custom written dialogic based solution.
The issue is that their dialogic solution can read
callerid from incoming calls, even if the callerid is blocked.
I have read before that Asterisk can do this, and
they want me to m
I hope the staff are not answering the question with their middle digit.
That would be rude.
PaulH
- Original Message -
From: "Ronald Wiplinger" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 01, 2006 12:45 AM
Subject: [Asteri
Sounds great - be there and be square.
PaulH
> jurgen <[EMAIL PROTECTED]> wrote:
>
> Hi all,
>
> Come one come all! We're having the next Asterisk evening at the
> Fujitsu Centre for Excellence! This is Fuji's state of the art
> show-off centre - they're promising lots of interesting toys to
Base your internal dialplan around Dial(Agent/xxx) rather than DIAL(SIP/xxx)
for calling extensions.
later,
PaulH
> Ben Ferguson <[EMAIL PROTECTED]> wrote:
>
> So I'm trying to set up queues and agents and am trying to figure out
> the
> best way to set up what I need to do. And what I need
Or put it on one of your DID's, I suppose. (if you have E1/T1)
PaulH
- Original Message -
From: "Sean Cook" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, January 27, 2006 8:16 AM
Subject: Re: [Asterisk-Users] checking voicemail via trunk
Move to PRI - it will be much more fun than working
with analog.
PaulH
- Original Message -
From:
Cisco - Kameko
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 6:17
PM
Subject: [Asterisk-Users] Digium
hardware
Hello,
I w
What error do you get when trying to call the SIP
phones?
PaulH
- Original Message -
From:
abc def
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 11:58
PM
Subject: [Asterisk-Users] Help with sip
setup because can't receive calls
Why would you want to get line 16?
PaulH
- Original Message -
From: "Dane Reugger" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, January 25, 2006 4:09 AM
Subject: Re: [Asterisk-Users] Hardware recommendations
> If you have 16 call a
Hmmmdo you mean that the system needs 4 lines? Or that you need a phone
that can make 4 concurrent calls?
PaulH
- Original Message -
From: "Dane Reugger" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 24, 2006 4:32 PM
Subject
Use different prefixes for different outgoing
calls?
(I know that's a nuisance though)
PaulH
- Original Message -
From:
Franklin Webb
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 24, 2006 7:34
AM
Subject: [Asterisk-Users] Fw: setting
outgoi
Needing a 4 line phone is going to decrease your choices of phones.
Why do you need 4 lines?
PaulH
- Original Message -
From: "Dane Reugger" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 24, 2006 2:45 PM
Subject: [Asterisk-Users] Hardware recommendations
> We would like to test Aste
Sorry about this - I hit send by accident while I was still writing the
email.
Pretend it never happened.
PaulH
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 24, 2006 7:26 AM
Subject: Re: [Asterisk-Us
Cute?
But it can use LDAP...
PaulH
- Original Message -
From: "Ben Klang" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 24, 2006 3:58 AM
Subject: [Asterisk-Users] Announcing PodMail 1.0 (GPL)
> Hello Asterisk Community.
>
>
Sadly, most of the phone manufacturers do not understand Southern Hemisphere
daylight savings.
Don't know why, but they just don't.
PaulH
> Rod Bacon <[EMAIL PROTECTED]> wrote:
>
> I've posted this to SNOM, but was wondering wheter anyone here has
> issues with
> SNOM 190 phones not showing
Quick question - what is the point of speex? Do we really need it as an
option?
PaulH
- Original Message -
From: "stevanus" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, January 19, 2006 3:37 PM
Subject: [Asterisk-Users] speex in aster
This would be great to have working!
PaulH
Blackburn, Vic
- Original Message -
From: "James Harper" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, January 19, 2006 3:08 PM
Subject: [Asterisk-Users] SMS to fixed phone line
> Telstra (A
Point taken.
At $1300 per month it really isn't worth it.
PaulH
- Original Message -
From: "Tim Litwiller" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, January 17, 2006 4:41 PM
Subject: Re: [Asterisk-Users] I need feed back on how an
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