Re: [Asterisk-Users] Asterisk Administration and Management requirements (splinter from $200 AMP bounty thread)

2004-11-12 Thread JAMES BOTHAM
Hi there,

I agree with Greg and also with the documentation
group, we are all great at bitching about * (I know I
have done a lot of it but thats because UK and support
for us is minimal or so it feels) we need to unite,
the only reason Microsoft are so popular is because it
take 2 minutes to install and applications are usually
it is easy to configure (coming from Windows to Suse
was quite easy due to YAST but then from Suse to
Fedora Core is a nightmare thank god for web min)
users a nd administrators don't want to be editing
conf files to do the smallest thing i.e. create a
dialplan  thats a nightmare, coming from an Avaya
background (although it has been a year since i
touched an Avaya INDeX) you could create powerful and
effective dial plans completely graphically it was so
easy anybody could do it. Although all system
administration was done through a menu driven console
and that was really simple to use. We need to take the
good from other systems and merge this to Asterisk.
Also we have to document it. theres no point in
writing the code if nobody can use it.

I would like to offer my skills to the production of
this I can document, bug test and am great at user
interface design I come from a software house
background which I can also utilise to get this
project off the ground.

I suggest that we all meet in a chat room to create
some form of a project map and get this off the
ground.


Cheers

James

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RE: [Asterisk-Users] $200 AMP documentation bounty

2004-11-12 Thread James Botham
Title: RE: [Asterisk-Users] $200 AMP documentation bounty





I would be interested in helping create this, I have extensive knowledge in web interface design and currently work for a software house, I can help with interface design, documentation and bug testing. I can read code but am not the greatest at creating code in C. I would be interested in starting/helping in an interface project as this is the greatest stumbling block for asterisk and will stop it from being commercially viable.

-Original Message-
From: Gregory Junker [mailto:[EMAIL PROTECTED]]
Sent: 12 November 2004 16:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] $200 AMP documentation bounty



How about this. It's obvious that there is a strong demand for a clean, 
portable, easy-to-use Asterisk manager application. Apparently those 
that exist currently fall short in one or more areas, or we wouldn't be 
having this discussion.


Why don't we compile a list of features that people want their manager 
app to do for them, and finally do one that works, is easy to install, 
and is intuitive? I'd be happy to manage the project, help code it, 
design it, whatever it takes. It'll give my decade or so of software 
engineering experience something to do.


I'll start off with the obvious:


- The app must allow the administrator, either graphically or via menu- 
and dialog-driven GUI, to add, remove and otherwise edit extensions 
within the Asterisk PBX software.


- The app must allow the administrator, either graphically or via menu- 
and dialog-driven GUI, to add, remove and otherwise edit users within 
the Asterisk PBX software.


- The app must allow the administrator, either graphically or via menu-
and dialog-driven GUI, to add, remove and otherwise edit channels within 
the Asterisk PBX software.


If anyone is interested in helping continue to describe what you want 
from an Asterisk system manager, I am perfectly willing to do my best 
and my part to make it happen.


Greg


dean collins wrote:
> http://sourceforge.net/projects/amportal/
> 
> AMP is this super manager that was set up a few weeks ago, basically it
> ties together about 3 or 4 other programs and presents it as a nice GUI
> display..
> 
> Lol, in other words for newbies forget it, I've tried like 3 or 4 times
> to install it and their documentation just plain sucks, so I'm prepared
> to pay anyone $200 that can write a step by step guide and then this can
> be posted online to help out others.
> 
> 
> Cheers,
> Dean
> 
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]] On Behalf Of Andrew
> Kohlsmith
> Sent: Friday, November 12, 2004 9:42 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] $200 AMP documentation bounty
> 
> On November 12, 2004 09:27 am, dean collins wrote:
> 
>>There is a $200 bounty for helping document a step by step guide to
> 
> AMP,
> 
>>anyone on this list interested in making easy money feel free to
> 
> contact
> 
>>me.
> 
> 
> What is AMP?  Asteirisk Manager Panel?  
> 
> (yes I realize this pretty much shows I am not going after the bounty,
> :-)
> 
> -A.
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RE: [Asterisk-Users] Caller ID for Japan?

2004-11-12 Thread James Botham
Title: RE: [Asterisk-Users] Caller ID for Japan?





Surprise, surprise you only have compatibility if you live in the US everybody needs to remember this...


-Original Message-
From: Isamar Maia [mailto:[EMAIL PROTECTED]]
Sent: 12 November 2004 13:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Caller ID for Japan?




Yes. It is possible. But a driver was not implemented for that yet.


Isamar



On Fri, 12 Nov 2004, Kuniyoshi Murata wrote:


> Hi,
>
> Does anyone know if it's possible to make Asterisk's Caller ID function to
> be compatible with Japan's "Number Display" system?
>
> TIA
> Kuni
>
> --
> Kuniyoshi Murata.iChat/AIM:macwebcaster
> English-Japanese Interpreter mailto:[EMAIL PROTECTED]
> Macintosh Webcast Specialist    http://www.macwebcaster.com
>
>
>
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RE: [Asterisk-Users] Echo - UK Impedance problem with X100P?

2004-11-12 Thread James Botham
Title: RE: [Asterisk-Users] Echo - UK Impedance problem with X100P?





I am not willing to spend money on a system where even the most basic things are incompatible with everybody but the US, this is just stupid I am using the X100P in a small office so that we can have a small PBX, seems daft having to upgrade my hardware just because of another US oversight. I am using a Gigabyte Titan 4 Pro motherboard and also tried it on another PC with an Intel motherboard and it had the same affect it could be Zapata.conf but pin pointing this is getting stupid I might invest in a commercial PBX as this is wasting so much of my time, its a good idea but saying the product is in its infancy is an overstatement.

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]]
Sent: 12 November 2004 13:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo - UK Impedance problem with X100P?



> I have an X100p interface (clone).  The system works fine but I get
> echo to a level where the system is all but unusable for IP  PSTN.  I
> seem to remember reading somewhere that the UK line impedance is 
> different from the default compile and needs changing.   I have
> Wikied etc, but found nothing yet.


The x100p (and presumably the clones) have an integrated circuit on the
board that was manufactured for use in the US with 600 ohm pstn lines.
The chip cannot be changed to any other impedance. However, there can
be many different sources for the echo and impedance matching is only
one of them. Others include:
 - incorrect * zapata.conf parameters
 - poorly engineered motherboards (eg, poor PCI bus, interrupt latency)


For zapata.conf, try something like:
echocancel=yes
echotraining=800


For poorly designed motherboards, there is no consolidated list of
which ones are good/bad so you're left with trying another one on your
own to see if it impacts the echo.


The TDM04B digium card (as an example) does not use that same pstn
chip, but rather another one from the same chip manufacturer. That
chip does have support for something like 18 different "country"
telco standards.




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RE: [Asterisk-Users] Echo - UK Impedance problem with X100P?

2004-11-12 Thread James Botham



Hi,
 
This 
sounds like the problem I have as well, the quality of the T100p is awful i get 
so much echo and as Chris says, if there is a solution to this I would love to 
know I have tried adjusting the RX, TX gain which sorta helpdesk but the call 
quality is still quite poor. If there is something we need to change before 
compiling then this needs to be documented.
 
Any 
help would be appreciated.
 
Thanks

  -Original Message-From: Chris Blunt 
  [mailto:[EMAIL PROTECTED]Sent: 12 November 2004 
  12:52To: [EMAIL PROTECTED]Subject: 
  [Asterisk-Users] Echo - UK Impedance problem with X100P?
  
  Hi, 
   
  I have an X100p interface 
  (clone).  The system works fine but I get echo to a level where the 
  system is all but unusable for IP – PSTN.  I seem to remember reading 
  somewhere that the UK line impedance is different from 
  the default compile and needs changing.   I have Wikied etc, but 
  found nothing yet.
   
  Any pointers 
  appreciated.
   
  Regards
   
  Chris 
  Blunt
   
  --
   
  SIP: 
  [EMAIL PROTECTED]
   
   
   



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RE: [Asterisk-Users] Amount of time asterisk take to pickup incom ing call on ZAP interface

2004-11-10 Thread James Botham
Title: RE: [Asterisk-Users] Amount of time asterisk take to pickup incoming call on ZAP interface






Hi,


Set callerId to No in Zaptel.conf should do the trick, although im in the UK and it answer it immediately because of the patch i am running.

Hope this helps


James
-Original Message-
From: Sophus [mailto:[EMAIL PROTECTED]]
Sent: 10 November 2004 13:55
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Amount of time asterisk take to pickup
incoming call on ZAP interface



Hi,


I understand now that asterisk is waiting for callerid info, is it
possible to stop asterisk waiting for this and just answer
immediately?


cheers
Adam
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[Asterisk-Users] UK CID patch and version 1.0 CVS build

2004-11-09 Thread James Botham



Hi 
there,
 
Not seen this error 
message before has anybody else, my perfectly working asterisk system (albeit no 
caller ID) has now stuffed up I was on the CVS head as of about a week ago but 
installed a UK caller ID patch so that I can get caller ID (no problems appeared 
to patch perfectly with no errors) after recompiling asterisk libpri and zaptel 
I now get the following error at the console and then it bombs out Any ideas 
??? I wish Mark and company would just incorporate this in to Zaptel its a pain 
in the A** to us UK users.
 
 
 [pbx_loopback.so]Nov  9 13:54:13 
WARNING[3593]: loader.c:248 ast_load_resource: 
/usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: 
pbx_substitute_variables_varsheadNov  9 13:54:13 WARNING[3593]: 
loader.c:429 load_modules: Loading module pbx_loopback.so 
failed!
James BothamClient 
Support ConsultantComputer Software Group plcwww.computersoftware.com- 


TALENT 
Sport Europe's first 'paper' Smart tickets for Reading FC & London Irish RFC www.computersoftware.com/smartticket.pdfTALENT CRM 
Electrolux Outdoor Products improve customer service with ROI in 6 
months! www.computersoftware.com/eop.pdf
_
 



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RE: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3

2004-11-05 Thread James Botham
Title: RE: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3





Patrick,


Thanks for this info its working fine now, I have added this info to the voip-info site for anybody else experiencing it.

Thanks


James


-Original Message-
From: Patrick Conroy [mailto:[EMAIL PROTECTED]]
Sent: 05 November 2004 12:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3



> I have the following error once I have installed asterisk on my new Fedora
> Core 2 test 3 box with a T100P, compiled fine no problems after using the
> info on voip-info.org but when i run ztcfg i get this error message: 
>   
> Notice: Configuration file is /etc/zaptel.conf
> line 143: Unable to open master device '/dev/zap/ctl'
>  



I had the same problem when I tried to install zaptel on FC2.  I
resolved it by commenting out the following lines:


ifeq ($(DYNFS),)
else
    @echo " Dynamic filesystem detected -- not creating device nodes"
    @echo " If you are running udev, read README.udev"
endif


from the zaptel Makefile (lines 233, 253-256 as of this morning's Makefile).


Hope this helps,
Patrick
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RE: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3

2004-11-05 Thread James Botham
Title: RE: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3



Ignore 
my last comment its worked hadn't commented it correctly.
 
Apologies 
 
James

  -Original Message-From: James Botham 
  [mailto:[EMAIL PROTECTED]Sent: 05 November 2004 
  12:27To: 'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: RE: [Asterisk-Users] Zaptel Issue in Fedora 
  Core 2 test 3
  I have done as below now I get a new error message when trying 
  to make clean or make install it says 
  Makefile:230: *** missing rule before commands.  
  Stop. 
  In the build I have these lines were slightly different the 
  ifeq ($(DYNFS),) is on line 230. I take it you didn't comment out the lines 
  below as these look like they are creating the device nodes.
  Any ideas 
  Thanks 
  James 
  -Original Message- From: 
  Patrick Conroy [mailto:[EMAIL PROTECTED]] 
  Sent: 05 November 2004 12:04 To: 
  Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 
  3 
  > I have the following error once I have installed asterisk 
  on my new Fedora > Core 2 test 3 box with a T100P, 
  compiled fine no problems after using the > info on 
  voip-info.org but when i run ztcfg i get this error message: >   > Notice: Configuration 
  file is /etc/zaptel.conf > line 143: Unable to open 
  master device '/dev/zap/ctl' >  

  I had the same problem when I tried to install zaptel on 
  FC2.  I resolved it by commenting out the 
  following lines: 
  ifeq ($(DYNFS),) else     @echo " Dynamic 
  filesystem detected -- not creating device nodes"     @echo " If you are 
  running udev, read README.udev" endif 
  from the zaptel Makefile (lines 233, 253-256 as of this 
  morning's Makefile). 
  Hope this helps, Patrick 
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RE: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3

2004-11-05 Thread James Botham
Title: RE: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3





I have done as below now I get a new error message when trying to make clean or make install it says 


Makefile:230: *** missing rule before commands.  Stop.


In the build I have these lines were slightly different the ifeq ($(DYNFS),) is on line 230. I take it you didn't comment out the lines below as these look like they are creating the device nodes.


Any ideas


Thanks


James


-Original Message-
From: Patrick Conroy [mailto:[EMAIL PROTECTED]]
Sent: 05 November 2004 12:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3



> I have the following error once I have installed asterisk on my new Fedora
> Core 2 test 3 box with a T100P, compiled fine no problems after using the
> info on voip-info.org but when i run ztcfg i get this error message: 
>   
> Notice: Configuration file is /etc/zaptel.conf
> line 143: Unable to open master device '/dev/zap/ctl'
>  



I had the same problem when I tried to install zaptel on FC2.  I
resolved it by commenting out the following lines:


ifeq ($(DYNFS),)
else
    @echo " Dynamic filesystem detected -- not creating device nodes"
    @echo " If you are running udev, read README.udev"
endif


from the zaptel Makefile (lines 233, 253-256 as of this morning's Makefile).


Hope this helps,
Patrick
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[Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3

2004-11-02 Thread James Botham



Hi,
 
I have the following 
error once I have installed asterisk on my new Fedora Core 2 test 3 box with a 
T100P, compiled fine no problems after using the info on voip-info.org but when 
i run ztcfg i get this error message:
 
Notice: Configuration file is 
/etc/zaptel.confline 143: Unable to open master device 
'/dev/zap/ctl'
The 
cards appear to load fine when i modprobe them  no errors appear but 
Asterisk cannot use them when I make a call to the line in question asterisk 
doesn't acknowledge them, the config I am using is off my old machine (identical 
version same hardware and card just new OS) no errors on its startup euitither 
it seems to accept the config fine. | suspect that the channel has possibly 
changed but without ZTCFG i don't know how to check, does anybody have any ideas 
on a) what this message means and B is there anyway to see the channel the 
device is using.
 
Cheers
James BothamClient 
Support ConsultantComputer Software Group plcwww.computersoftware.com- 


TALENT 
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_
 



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RE: [Asterisk-Users] T100P Caller ID UK

2004-11-01 Thread James Botham
Title: RE: [Asterisk-Users] T100P Caller ID UK





I agree, 


When will all the Americans realise that the planet is much bigger than just them, (not being insulting, I think the US is a fantastic country) The technology works over here great but just little features like that will stop * from being a platform in a country these are such basic PBX functions that are required by all end users. And its bad enough paying £2.50 a month for it and not being able to use it.

The patch we had in Mantia works great why don't they just apply it and have done. 


Cheers


James


-Original Message-
From: Jon Lawrence [mailto:[EMAIL PROTECTED]]
Sent: 01 November 2004 17:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T100P Caller ID UK



On Monday 01 November 2004 14:11, Alex Barnes wrote:
> I have to agree take that is a big slap in the face to the UK based
> customers/users.
>
> All I can assume is Digium don't need our money / support.
>
Indeed.
I couldn't careless about bloat in the driver - at the end of the day, the 
config process could be altered so that the bloat is only there for those 
that need/want it. I need it to work end of story.
If a rolling buffer is the only way then so be it.
There isn't a digium solution to connect to POTS lines in the UK other than 
X100P's, and I for one can't live without callerID - I'm even considering 
going across to ISDN so that callerID continues to work with future * 
versions.


I have the patches against version 1 for X100P callerID, will upload them to a 
server at some point - when I find where I've put them.


Jon
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[Asterisk-Users] T100P Caller ID UK

2004-11-01 Thread James Botham



All,
 
Has anybody had any 
luck using the diffs for UK caller ID on the latest CVS versions of asterisks, 
if so does anybody have a diff that currently works against version 1 or the 
current CVS head. Quite annoying the comments on Mantia about adding facility 
for us peeps in the UK to have caller ID with the T100P as adding bulk to the 
Zaptel driver... 
 
Thanks
James BothamClient 
Support ConsultantDDI: 01270 
613 802Computer Software Group plcPepper House, Market StreetNantwich, Cheshire, CW5 5DQT: 01270 
613 800F: 01270 613 801www.computersoftware.com- 


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RE: [Asterisk-Users] NetworkWorld article on Open Source Telephon y

2004-06-09 Thread James Botham
Title: RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony





I agree, any platform suffers when it is extremely difficult to implement. What we need is an interface that does everything we need and shows what asterisk is capable of, a lot of features will go unused because you might not know the exist unless you hunt them down in the source or conf files. 

I trained on an Avaya INDeX switch it had a complex console but was laid out in a structured way a java console that allowed you to issue changes to the system would sort it out and would shy away from a GUI that is restrictive, things such as call flows etc... were done in a very simple GUI, although simple this GUI could do very complex stuff. Perhaps we need a suite of tools for it each specialising in an area and linked together via a configuration database.

-Original Message-
From: Chris Bond [mailto:[EMAIL PROTECTED]]
Sent: 09 June 2004 14:42
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] NetworkWorld article on Open Source
Telephony



> The power of asterisk comes from its method of config. If one wraps it 
> with a GUI one will inherently limit the flexibility.


> Then since the GUI is what gets 'seen' people ~may~ take the lack of 
> flexibility or even just the look and flow of the GUI to be a reflection 
> on the power of Asterisk.


But if it was an official addon from the cvs tree (similar to the voicemail
cgi stuff), it would make take-up a lot easier =) 


That way you wouldn't make people "stuck" to one GUI, if they don't want it
they don't need to check it out. Its just at the moment, you've got sub
projects for lots of different GUIs, what needs to happen is someone to
consolidate what's out there and bring it all into one official project.  It
makes sense that the GUI becomes a web one, then it can run on a number of
web browser platforms.


Kind Regards,
Chris Bond


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