Re: [Asterisk-Users] PHP/AGI Problem

2005-05-26 Thread Matthew Asham
On Thu, 2005-26-05 at 17:59 -0400, Michael Stearne wrote:
 On 5/26/05, Benjamin West [EMAIL PROTECTED] wrote:
  Michael,
  The version, in the context of Jon's problem, was irrelevant.  Jon's
  problem was due to a small bug in his code, and not related to PHPAGI.
 
 Yeah.  I was just wondering what version people were using since 2.0
 hasn't been formally released yet.  I spoke to the developer and he
 suggested 2.0.
 

phpagi-2.14 has now been released, this is the first official release
for the 2.x series but should be stable.

There is also a compatibility module in CVS to support older 1.x apps.

You can snag it from http://phpagi.sourceforge.net/

-- 
Matthew Asham - the B.C. Wireless Network Society
www.bcwireless.net - +1 604 484 5289 x1006



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Re: [Asterisk-Users] PHP/AGI Problem

2005-05-26 Thread Matthew Asham
On Thu, 2005-26-05 at 21:23 -0400, Michael Stearne wrote:

 Thanks for the efforts Matt.

David Eder is the one who deserves the praise, he's done wonderful work
on version 2.  

Matthew



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[Asterisk-Users] NuFone problems to non-na numbers

2005-04-19 Thread Matthew Asham
Is anyone else having problems with Nufone dialing international (non
NA) numbers? 

Pretty much every intl number dialed comes up with a voice intercept
saying the call could not be completed as dialed.  Tried it with two
separate accounts, and the numbers themselves work from the local
telco.  

The problem appears to have started within the last few days (and yes I
have emailed [EMAIL PROTECTED], just wondering if we're the only ones
having the problem).

Matthew


-- 
Matthew Asham - the B.C. Wireless Network Society
www.bcwireless.net - +1 604 484 5289 x1006

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Re: [Asterisk-Users] Text Messaging or AIM

2005-03-13 Thread Matthew Asham
On Sun, 2005-03-13 at 11:14, Peter Svensson wrote:

 Whatever gave you that idea? Most operators have an interface allowing 
 reception of sms:es over internet. The protocols may be strange (they are) 
 and the pricing models vary greatly, but there are many receive interface 
 to sms:es.

I've been wondering about this for some time, is there a common product name 
for 
this service?  



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Re: [Asterisk-Users] Digium Hardware in Canada

2004-12-14 Thread Matthew Asham
On Tue, 2004-12-14 at 11:09, Adi Linden wrote:
 I am looking for a supplier of Digium hardware in Canada. Any suggetions?
 

Other coast, but http://www.netvoice.ca/digium/


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Re: [Asterisk-Users] SysMaster and GPL Violation (lets think before we

2004-11-14 Thread Matthew Asham
What's a top post? 

oh and, me too!


On Sun, 2004-11-14 at 18:15, [EMAIL PROTECTED] wrote:
 Does that mean I have to put my posts there when I'm PMSing?
 
  On Sun, 14 Nov 2004 17:30:23 -0800, Bruce Ferrell wrote:
 
 Only if we can move the top-post discussion there too
 
 Gary wrote:
  Hi folks,
 
  Might I propose a new mailing list ??
 
  Asterisk-bitch
 
  Thus discussions such as the one with this topic could be moved to it
  rather than clutter up an already very busy list.
 
 
  All those in favour ?
 
 
 Only if we can move the top-post discussion there too
 
  Sure,
  any of those types of debate.
 
  In fact, i just wished people would NOT use the list for debates !!
  .
 
 
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Re: [Asterisk-Users] Enhancing list quality?

2004-11-02 Thread Matthew Asham
people complaining about list etiquette...


On Tue, 2004-11-02 at 04:45, Patrick wrote:
 Hi all,
 
 Following up on Steve Szmidt's post, have we come to a point where
 something should be done about the deteriorating signal to noise ratio
 on the asterisk-users list?
 
 Similar to other fast growing successful FOSS projects/communities, the
 Asterisk community is no exception when it comes to growing pains.
 Anyone who has been subscribed for quite some time surely has noticed:
 
 * the lazy top posting
 * asking too simple/obvious questions that are found on voip-info.org,
   the list archives, google and the docs that comes with the source
 * asking totally unrelated questions (the Sparco example)
 * asking why it doesn't work without giving any config info
 * the GPL license, Windoze vs. Linux, You are clueless! No you are!
   discussions/flame wars that go on and on and on
 * the lack of proper trimming of posts
 * the html email sent time after time in spite of requests to 
   turn it off
 * the huge signatures that imho are a form of spam or just a waste of
   bandwidth
 
 Some suggestions to hopefully improve things:
 
 1) change the Welcome to the asterisk-users list email to contain:
 
 - before sending email to more than 8000 people, search voip-info.org,
   the mailing list archives, google and the docs in the source
 - do not top post
 - add as much config info as possible
 - trim long posts (especially unnecessary sigs)
 - posts should be related to Asterisk
 - be nice, polite, courteous. keep flames off list
 - your sig should not be a billboard for your products/services (spam)
   if you want to advertise your services, use asterisk-biz
 - do not send html mail to the list. It will get rejected or bounced
   or whatever (see item 3)
 - do not use unnecessary capitals, exclamation marks and question
   marks (don't be loud)
 - do not mail questions to asterisk-dev that are not related to the
   actual development of asterisk
 - do not crosspost your question to asterisk-users and asterisk-dev
 - do not resend your email after 4 hours because you have not
   received an answer. Join #asterisk on Freenode irc if you require a
   possibly speedier answer
 
 2) post this info periodically to the list to remind people
 
 3) strip all html or bounce it back to the sender telling them to send
normal email, not html email. Add the list's netiquette too, just to
make sure. Afaik mailman is capable of stripping html from email
 
 4) ask the community to respond to people ignoring these reasonable
requests and gently point them in the right direction
 
 Feedback welcome.
 
 Regards,
 Patrick
 
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Re: [Asterisk-Users] common numbers ?

2004-10-22 Thread Matthew Asham
http://www.google.ca/search?hl=enq=vertical+service+codesbtnG=Google+Searchmeta=

On Fri, 2004-10-22 at 01:57, [EMAIL PROTECTED] wrote:
 hi,
 
 Can someone point me to a list of a common numbers used for different
 functions ex. callparking,forwarding etc...
 I can thin of my own but want to know is there some standard wich is
 good to follow.
 
 tia
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[Asterisk-Users] viewing fax tiffs?

2004-09-23 Thread Matthew Asham
Hello,

I have spandsp setup to accept incoming faxes and receiving tif files
via Email.  

Using tiff2pdf, or tiff2ps -a2 or even tiffsplit, the last page of the
fax is cut off and the quality of the text looks squished.

I figure it's a tiff parsing thing, as opposed to a problem with my
spandsp installation (heh).  

Has anyone experienced the same thing, or can anyone recommend a
proper way to convert fax'd tiffs to another format or even a
multi-page tiff viewer for Linux?

Thanks

Matthew



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Re: [Asterisk-Users] e164.org

2004-05-22 Thread Matthew Asham
On Sat, 2004-05-22 at 18:08, Tony Hoyle wrote:
 Simon Dorfman wrote:
 
  I wonder if someone can help me understand this.  Let's say I configure my
  asterisk box to use e164 and then I try to call a phone number in Germany.
  I'm in the U.S.A.  So if the number I'm calling in Germany is registered in
  e164's dns, would my call be routed directly via their voip provider?  Or
  directly to their asterisk box?  And would it be free?
 
  From the looks of it, they're just a directory... it looks like their not 
 running asterisk themselves.

It's a DNS root, that Asterisk (via the EnumLookup application) can
use.  The EnumLookup() application will resolve the number to a dial()
channel.  

ie:

; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup

To get * to resolve against e164.org, add:

search = e164.org

to /etc/asterisk/enum.conf.


So yes Simon, if you called someone in Germany and it was the zone, your
call would be switched over the 'net.  If not, you could drop it to
NuFone or some other carrier.

 They use something called EnumLookup which I guess is some kind of 
 plugin/script.  If the number you're calling is in their database, it calls 
 the VOIP number directly, otherwise it calls the POTS number

Or whatever else your dial plan wants to do.

Matthew


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Re: [Asterisk-Users] e164.org

2004-05-22 Thread Matthew Asham
You know, sleep deprivation cause people to do dumb things.  The example
I pasted was hastily pasted and renumbered, 

 exten = _1NX,6,Playback(enum-lookup-failed)
  exten = _1NX,7,Hangup

are actually:

exten = _1NX,103,Playback(enum-lookup-failed)
exten = _1NX,104,Hangup


Duane wrote up some more detailed examples at
http://www.e164.org/config.php.

Sorry for not proofing that when I posted it.  I'll go sleep now.

On Sat, 2004-05-22 at 18:46, Tony Hoyle wrote:
 Matthew Asham wrote:
 
  ; north america enum
  exten = _1NX,1,Playback(doing-enum-lookup)
  exten = _1NX,2,EnumLookup(${EXTEN})
  exten = _1NX,3,BackGround(enum-lookup-successful)
  exten = _1NX,4,Dial(${ENUM},30,tr)
  exten = _1NX,5,Hangup
  exten = _1NX,6,Playback(enum-lookup-failed)
  exten = _1NX,7,Hangup
  
 Interesting.. how does it know to go to '6', or does it just jump +4
 on failure?
 
 That reminds me I seriously need to restructure my extensions.conf... there's 
 no way currently I could add anything like that without major surgery (only 
 discovered the 'local' target this afternoon so I have everything copied/pasted).
 
 Tony
 

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[Asterisk-Users] IAX/SIP in 604?

2004-04-02 Thread Matthew Asham
Hello,

I hate to ask here, but..

Does anyone know of an IAX/SIP DID provider in Vancouver, British
Columbia?  I'm looking for a voicepulse isk service, one DID with
standard calling features and some sort of long distance package.  

I've looked around on voip-info.org's list of VoIP providers but so far
I haven't found one that offers a 604/778 number.

Thanks

Matthew

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[Asterisk-Users] How does Asterisk use CPU?

2003-11-28 Thread Matthew Asham
Hello,

I'm trying to figure out what portions of Asterisk need a lot of CPU
time.

I thought I read somewhere that a Dual P4 2.something will support
approximately 80 calls.  Is this based on calls that Asterisk is
actively doing voice processing for (say, Zap channels and voicemail)?

Would a SIP client going through Asterisk and out an IAX channel be
CPU intensive if I kept the codec the same throughout the path?

I'm probably not asking very clearly, it's awefully late (err,
early) but any pointers would be greatly appreciated.

Thanks

Matthew

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Re[2]: [Asterisk-Users] How does Asterisk use CPU?

2003-11-28 Thread Matthew Asham
Thanks Steve, that pretty much answers what I wanted to know.

I was asking more out of general than for a specific deployment, but
if I do have further questions I'll be sure to elaborate. :)

Matthew

Friday, November 28, 2003, 6:41:23 AM, you wrote:

 On Fri, 2003-11-28 at 06:51, Matthew Asham wrote:
 Hello,
 
 I'm trying to figure out what portions of Asterisk need a lot of CPU
 time.

 Mostly codec translations.

 I thought I read somewhere that a Dual P4 2.something will support
 approximately 80 calls.  Is this based on calls that Asterisk is
 actively doing voice processing for (say, Zap channels and voicemail)?

 I think that thread would have been more specific to a task, or just
 blanket overkill.

 Would a SIP client going through Asterisk and out an IAX channel be
 CPU intensive if I kept the codec the same throughout the path?

 If the codec is the same, then all that is being done is reformating the
 control protocol. If there is any codec translations, then you would run
 into some extra overhead.

 I'm probably not asking very clearly, it's awefully late (err,
 early) but any pointers would be greatly appreciated.

 Maybe you would do better to ask more pointed what YOU need help with.
 If you need help with specing out a machine for your installation, then
 start specing your installation so we can help. 

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