Sean,
The dealer is talking utter bol**cks...
You simple need the SIP firmware from Cisco and a TFTP server running on a pc. I have
a Cisco 7940 G running happily with an * box. Not only that but I'm able to happily
use the XML interfacing on the phone for things like a phone directory and
hi,
not sure is anyone is aware, but I found a perl module that makes interfacing with a
cisco 79x0 phone a breeze
http://www.cpan.org/modules/by-module/Cisco/
Though it might be of some use...
Andy
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Personally I'd use ssh rather than telnet
Andy
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On 15/08/2003 at 12:21 Steve Lane wrote:
I am having problems trying to run asterisk from a telnet session. I am
able to su to root and the command asterisk does not work. Any ideas why
this may be
Is this not just a case of a new entry in sip.conf
EXTERNIP = external IP
with the code for the contact header modified to use it (if present). Then the
external firewall is set to forward the rtp and 5060 to * ..
I know many people either have sip aware firewalls (as i do) or their * box
Errm, no...
does that mean you'll personally check to see if my line is busy or not ;P
will try it now...
Andy
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On 14/08/2003 at 09:58 Martin Pycko wrote:
Did you try BUSYDETECT_MARTIN in asterisk/Makefile ?
regards
Martin
On Thu, 14 Aug 2003, Andy
FCC mode is for the US. CTR21 is for Europe - you even pasted the info
in your message!
Exactly, the question really is how do you change it?
modprobe wcfxo opermode=1
HTH
Andy
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I was pondering on this question, and have to agree, splitting mailing list just means
yet another list to join (since there may one day be something relavant) and filter
locally. What might appear to be a good solution is a privately run newsgroup on a
digium server eg news.digium.com the
Fabia,
The only numbers you should be able to dial from that config are
1945
1943
2999
and nothing else...
The entry under bogon-calls (isn't it bogus calls?) should read
exten = s,1,Congestion
rather that using the _. ...
HTH
Andy
*** REPLY SEPARATOR ***
On 10/08/2003
Hi,
when using multiple * boxes, there appear to be 2 choices as to how to go about
sharing cards and dialplans
1) using switch
2) using dial fail fall-through ie
exten = 1,1, dial(xyz)
exten = 1,2, dial(otherpbx/xyz)
As i see it switch could end up being recurrsive resulting in a wild ooc
It's just a proxy service like fwd it will work with asterisk... The phones they are
selling
with the deal are Grandstreams.
It's very likely that they just have been preloaded with their settings, and probably
point to their
own tftp server. simply create fake dns entries and a static route
On 13/08/2003 at 17:46 Dave Cotton wrote:
in the file wcfxo.c the following structure is initialised as below
which would suggest that FCC is wrong for France or pretty well all of
Europe.
errm,
FCC mode is for the US. CTR21 is for Europe - you even pasted the info
in your message!
See below
In fact the is is not required,
see below:
On 06/08/2003 at 15:50 Michael Robertson wrote:
The phones are completely preconfigured, but not locked in any way to
the SIPphone service. Owners are free to change any settings they want.
-- MR
Andy Powell wrote:
Hi,
Might seem an obvious
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On 14/08/2003 at 20:48 Richard Scobie wrote:
Andy Powell wrote:
FCC mode is for the US. CTR21 is for Europe - you even pasted the info
, Andy Powell wrote:
Hi Dave,
I have a similar problem, I tried using busydetect and busycount but
calls kept being dropped
at random intervals. It didn't seem to matter what i set the busycount
to. I guess it's a case
of deciding which is more important... You can also limit the length
I think this is a good idea but at least for FWD users can't they just use
the FWD proxy that is
designed to handle clients behind NAT with no special software on the
client. The ones that
allows even Windows Messenger to work behind NAT.
Sadly this doesn't work otherwise they'd all be using
Hi Mark,
Short of taking my board out of my * box is there any way to check what revision of
the TDM400P I have? It was purchased in May of this year. Is the pricing likely to be
the same or similar to the add-on FXS ports? Does this also mean that as we'd be able
to get away with not using
Mark,
if the capability for line reversal detection is in the hardware (X100P) then does
this mean that the detection of DTMF style caller-id as used in the following
countries would ber trivial? or am I hoping too much...
Finland, Denmark, Iceland, the Netherlands,India, Belgium, Sweden,
Hi Dan,
I use panasonic DECT phones, when plugged into a TDM20B (2 port FXS card from Digium)
I get caller id passed through (name AND number) although i can't get callerid via the
pstn at the moment (located in nl) i do get it for VoIP calls. Plus when a pstn call
comes in and there is no
of seconds when you pick-up
the phone, to know that you have a voice message waiting.
BR,
Dan
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 08, 2003 2:34 PM
Subject: Re: [Asterisk-Users] CallerID, DECT phones and ATA
Hi Dan,
I use
You also appear to have a big problem with your clock...
unless you are from the future.. in which case how are Glaxo stocks doing?
Andy
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Steve
I have to say that some listserv's do allow this .. at least he didn't reply to 20
messages with
REMOVE
in them
Andy
On 07/08/2003 at 10:10 Steve Meyers wrote:
On Thu, 2003-08-07 at 10:01, Justin Carlson wrote:
unsubscribe
Has anyone ever been on a mailing list where you could
Garry,
yes this is possible although it would end up being quite convoluted.
Essentially you could have a cron job that monitors your voicemail directory, or use
the perl manager interface to check the status. Once it has been established that you
have message(s) submit a .call file to dial
Hi Dan,
For a totally unrelated reason I did this today. * runs fine here
under VMware, athough I haven't stressed it at all.
Andy
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On 07/07/2003 at 19:07 Dan wrote:
Hi,
There is any experience using Asterisk with VMWare?
I think about installing a
Hi,
You just have to be a little patient...
try
http://www.automated.it/guidetoasterisk.htm
as a start, it might at least get you going with sip based stuff. I don't
like to particularly push the guide specifically because it's mine. I'd rather
you got it by recommendation... but hey,
WipeOut,
IIRC the
qualify=yes
directive in your iax.conf definition for the switch causes * to check to see
if the host is alive.
Andy
Is there a way to get asterisk to verify that the remote host is in fact
available before attempting the switch so that if it is unavailable the
local
Tim,
a good comprehensive answer to the question...certainly gave me a few things
to think about. I do have a few questions though, since I'm in Europe.
Has anyone in Europe set up something equivalent to what Tim suggested?
What sort of prices did it work out at?
How did you solve the
Dan,
The first question is :
is your voicemail in the default location or have you moved it to another disk?
if you do this you need to update the vm system link in the
/var/spool/asterisk directory eg:
vm - /home/asterisk/voicemail/default/
using ln -s new path vm
also make sure * has the
You could create a simple moh class that played a silent mp3 as a very low rate,
or even the occasional beepthen just use setmusiconhold,newclass
hth
Andy
On 27/06/2003 at 13:10 Derek Beaumont wrote:
I was playing with the agent application to see if I could get it to
work.
Everything
The X100P is modprobe wcfxo
The TDM40B is modprobe wcfxs
Andy
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On 27/06/2003 at 16:07 Steven P. Donegan wrote:
What is the module name for the TDM40B - I received my X100P and TDM40B
today (thanks Digium).
___
Mmm...
I don;'t know what else to try, I've had callerid turned on here but it doesn't work
at all... :(
Andy
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On 26/06/2003 at 13:02 Dan wrote:
There is nobody with an X100P in Europe having this issue related to the
PSTN Caller ID?
Please help!
I tell you what, just relpy to every message with the word remove
rather than actually reading the instructions.
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On 26/06/2003 at 13:30 cisb wrote:
REMOVE
- Original Message -
From: Peter Zeltins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Well, I can't get it to work with an MD110 PBX over here either (.nl). It
probably should work, but I never found how. Tried several options as
suggested by this list though..
--
From what I can gather the caller id in NL is similar to Denmark, it's just
a series of DTMF tones send down the
Oliver,
can you clarify how the gateways is supposed to be used,
I've tried calling the number from a PSTN line, the call is answered
and i get dialtone, I then try to dial my iaxtel number and just get
told that it's an invalid extension.. the 'error' occurs after dialing
17001 of my iaxtel
Anybody have the latest word on Snom's development? Last I had heard,
they were still working on compatibility with GSM. Firmware version
1.15e, which is what my Snom 100 automatically updated itself to, does
not work with GSM.
-Tilghman
I'm using 1.15u and it's a little better. Snom are aware
You can try Snom or Cisco...
Or get a TDM card and use an analog phone...
Andy
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On 25/06/2003 at 15:44 Chris wrote:
I've got Asterisk up and running nicely using a couple of different
softphones. Audio quality is suffering a bit due to the hardware
Andy, your update is
http://www.automated.it/guidetoasterisk.htm isn't it ?
yes, same place, just added some extra notes in there (they should be obvious)
HTH
Andy
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Hi,
I don't understand what i have to make and set to communicate with
external telephons SIP (Sjphone, X-lite, MS messenger ...)
Must i have a SIP provider subscription, how to integrate this
subscription with asterisk
Do you mean internally i.e. Sjphone, X-lite, MS messenger phones
on your
This is odd because I email all my users voicemail out and the ones that
don't clear the voicemail on the phone still get stutter tones. I had to
inform them of what to do, and then mass delete their voicemail to get
the stutter tone to stop. One user had almost 50 messages waiting.
I had this
On 20/06/2003 at 14:45 Wade Weppler wrote:
Same here. E-mail and MWI/Stutter tone work fine together.
if that attaching the file or just sending a messages without a file attached..?
Andy
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You have looked at festival.conf right? What's your
exten line line here's one of mine:
exten = 1021,1,Festival(mary had a little lamb)
Note the lack of quotes
hth
Andy
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On 19/06/2003 at 13:57 Chad Sawyer wrote:
I followed the directions I found in
On 17/06/2003 at 10:23 Rushowr wrote:
Is it possible to set up Asterisk without any of the cards? I'm
interested in setting it up for the company I work for, but I would like
to set it up and see how difficult it will be before I start having the
company spend a chunk on equipment.
Yes, you can
On 16/06/2003 at 10:26 DUSTIN WILDES wrote:
If this is through your Telco, they may have turned on the Callerid-Name
field along with your number.
I had mine turn on the Callerid-Name field for us.
No, not from my teleco, this is from * via the TDM card to the DECT phones
that's why it
Ok,
this has really freaked me out, but in a good way - sort of.. I've made no changes at
all to my system, save messing with ADSI. However this has nothing to do with ADSI.
The thing is all of a sudden my DECT phones have started reporting caller id, and not
just the number, the name too!
So is that one switch statement per installation or one per context.
ie can i have multiple switch statements each one applicable to a
different section in extensions.conf
Andy
On 13/06/2003 at 13:28 Martin Pycko wrote:
The idea of switch is for every box to know what it can reach locally.
Derek,
exten = 400,1,SetCallerID(${CALLERIDNUM})
You can use
${CALLERID}
${CALLERIDNAME}
${CALLERIDNUM}
Andy
On 13/06/2003 at 16:18 Derek Beaumont wrote:
I don't understand how or where I would use setcallerid.
I have tried to do
exten=400,1,Setcallerid,asreceived
but that doesn't seem to
Hi,
I've had a search through the archives and didn't find much. Is anyone using the
Monitor application? I have it working but there is a really big drawback. The files
are always called the same thing, which means if I make 2 calls one after the other
the first recording is lost. I half
Ahh, wonderful thanks...
Andy
On 12/06/2003 at 13:35 Pertti Pikkarainen wrote:
Check
http://www.loligo.com/asterisk/current/extensions.conf
and find macro called macro-record-on
There is at least one way described ( author is John Todd ).
--Pertti
Ok, thanks for the clarification
Shame it still doesn;t work for me :( maybe it only works
with US phones... anyone in Europe got this working?
Andy
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On 11/06/2003 at 21:55 Tilghman Lesher wrote:
On Wednesday 11 June 2003 19:10, Andy Powell wrote
Hi,
You need to change your settings in X-lite:
Display name : roseau
user name : 1000 --- this is wrong!
authorization user : same as user name
Password :
Domain/Realme : 192.168.0.2
SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty
to:
user name : roseau
(That should match the
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting together if you
like...
http://www.automated.it/guidetoasterisk.htm
I'd appreciate any feedback you have on it.. and if it helped
Andy
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On 06/06/2003 at 14:17 Tielman
On 06/06/2003 at 17:36 Patrick wrote:
Excellent stuff Andy. It was quite a disappointment that the document
stopped before explaining ..errr everything :) Look forward to learn how
to setup one-way conference and music on hold. Thanks for the guide so
far.
Regards,
Patrick
Glad it was of use
I've played with modifying the extensions.conf and h323.conf but don't
have things right. I keep getting a message on the
console:
ERROR[376849]: File chan_h323.c, Line 974 (setup_incoming_call): Call from
user 'Simon' rejected due to no default context
However I am unsure what this really
Sorry,
I might be being stupid, but I don't see what the problem is.
Following your example,
1. Secretary calls someone for the Boss
2. Other caller answers, Secretary asks other end to wait.
3. Secretary presses the flash button (or recall or whatever it's called on the phone)
4. Secretary
Nathan,
Get in touch with www.provu.co.uk ask to speak to Tim, and tell him you heard from me
(Andy Powell) that they had a deal running where you could get Snom 100's for 140
gbp...
HTH
Andy
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On 29/05/2003 at 12:44 nathan wrote:
Hi All,
What
Ok, so are you pressing # then hearing the word 'transfer' and dialing the exten to
transfer to?
Andy
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On 14/05/2003 at 11:34 Derek Beaumont wrote:
I am using a regular analog phone.
Derek,
What are you using to place the call? Snom Phone? Cisco
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