[on-asterisk] Astricon or Asterisk World?

2007-08-07 Thread John Lange
So I'm confused; Astricon is September 25 - 28, 2007, but Digium is also
sponsoring Digium Asterisk World October 30, 31 2007. From what I can
see this second conference is part of VON?

Just wondering if anyone has attended either of these in the past an has
any opinion on which would be more worthwhile?

John



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[on-asterisk] FALL VON 2007 - Time to hit road again

2007-08-07 Thread Bill Sandiford
Hello All:

Back in June I organized a small little road trip to the NXTcomm Telecom 
tradeshow in Chicago.  Reza and I told the group about the show at June 
meeting.  It was amazing.

Well the time has come to start planning again.  This time the show is the Fall 
VON (Voice on the Net) show which is being held in Boston from October 
29th-November 1st.  The exhibit hall is open on the 30th and 31st.  The VON 
show is mostly focused on VoIP (NXTcomm was all aspects of Telecom).  Here is 
the link to the info site on the show:

http://www.von.com/2007/boston/web/

I have attended this show for the past 3 years.  Although the show seemed to 
have shrunk a little bit last year, I have just looked at the floorplan online 
and it looks like this years show is back up to its previous size.  Here is the 
link to the floorplan

http://fp1.a2zinc.net/clients/fpvon/fall2007/public/fp.aspx

Of specific interest to our group is that this years Fall VON is hosting 
Digium/Asterisk World.
http://www.digiumasteriskworld.com/2007/boston/web/

From the floorplan it looks like there is at least 4 booths setup for 
Digium/Asterisk World, 20ft x 20ft, 20ft x 30ft, 75ft x 30ft, and a 
monstrous 95ft x 40ft.  Thats a total of over 7000 sq ft of space dedicated 
Digium/Asterisk World.

Not to mention the fact that once again most of my major vendors are there so I 
should be able to get us in to some good parties in the evenings.

So, calling all TAUG members.  Who is up for a road trip to Boston?  Let me 
know asap as I would like to get this one planned well in advance.  We could 
have a couple of different options for travel.  We could carpool in a van again 
(hopefully multiple vans if we have enough people).  It is about a 9 hour 
drive.  Also, as much as I enjoyed the 8.5 hour drive to/from Chicago with 
Norm, Todd and Reza (cough cough), JetBlue has really cheap non-stop flights 
from Buffalo to Boston for about 129 USD taxes included (about $140 CDN these 
days).  So, I'm somewhat leaning towards flying down, probably early on the 
30th (there is a 7:00am flight out of Buffalo that gets to Boston at 8:25am) 
and flying home early on the 1st.

These cheap flights will fill up fast, so we will need to book soon !!!

Any takers?

Bill Sandiford
Telnet Communications
905-674-2000 x100
[EMAIL PROTECTED]

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[on-asterisk] Checking call volume

2007-08-07 Thread David Steele
Hi,

Does anyone have any quick tips on checking the number of active calls on an
Asterisk box?  I think my PRI may be overloaded.

Also, does anyone have any recommendations for a program to analyze CDR data
- ideally just an excel spreadsheet with some smarts built in...

Thanks,
Dave.

-- 
___
David Steele

insert sig line witticism here


Re: [on-asterisk] Checking call volume

2007-08-07 Thread philip mullis
show channels and zap show channels works pretty good on the cli,
otherwise you can pull it from your cdr data. 

Regards,

Philip Mullis

On Tue, 2007-08-07 at 14:24 -0400, David Steele wrote:
 Hi,
 
 Does anyone have any quick tips on checking the number of active calls on an
 Asterisk box?  I think my PRI may be overloaded.
 
 Also, does anyone have any recommendations for a program to analyze CDR data
 - ideally just an excel spreadsheet with some smarts built in...
 
 Thanks,
 Dave.
 


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[on-asterisk] Interswitch call - driving me nuts!

2007-08-07 Thread David Cook
This has been driving me nuts and I hope it is just a forest/trees thing.

 

2 switches, both defined to each other. Site A calls Site B - works like a
charm. Site B calls Site A - I get authenticated, get the IVR, make my
selection, it says it is dialing the user's extensions then I get sip
channel (remote end) auto congesting.

 

Of course, all calls on-site work fine, remote to other sites works fine,
DID's from lesnet work fine, etc. Finally, I get a 400 error back from the
phone. (local on-site calls use macro-cacexten too).

 

-- Executing [EMAIL PROTECTED]:6] Dial(IAX2/SITEA_Server-1,
SIP/201|25|tw) in new stack

-- Called 201

[Aug  7 14:52:35] NOTICE[4747]: chan_sip.c:2809 auto_congest:
Auto-congesting SIP/201-19c309e0

-- SIP/201-19c309e0 is circuit-busy

  == Everyone is busy/congested at this time (1:0/1/0)

-- Executing [EMAIL PROTECTED]:7] Goto(IAX2/SITEA_Server-1,
s-CONGESTION|1) in new stack

-- Goto (macro-cacexten,s-CONGESTION,1)

-- Executing [EMAIL PROTECTED]:1]
Playback(IAX2/SITA_Server-1, all-circuits-busy-now) in new stack

-- IAX2/Advan_Server-1 Playing 'all-circuits-busy-now' (language 'en')

-- Executing [EMAIL PROTECTED]:2]
Hangup(IAX2/SITEA_Server-1, ) in new stack

  == Spawn extension (macro-cacexten, s-CONGESTION, 2) exited non-zero on
'IAX2/SITEA_Server-1' in macro 'cacexten'

  == Spawn extension (macro-cacexten, s-CONGESTION, 2) exited non-zero on
'IAX2/SITEA_Server-1'

-- Hungup 'IAX2/SITEA_Server-1'

-- Incoming call: Got SIP response 400 Out Of Order back from
192.168.1.202

-- Incoming call: Got SIP response 400 Out Of Order back from
192.168.1.201

 


Site A

Site B


Asterisk 1.2

Asterisk 1.4.9


Cisco 7960

Aastra 9133i


 

 


IAX.CONF

 


[SITEB_Server]

type=friend

host=dynamic

secret=xxx

context=from-cac

disallow=all

allow=ulaw

allow=gsm

[SITEA_Server]

type=friend

host=dynamic

secret=yyy

context=from-advan

disallow=all

allow=ulaw

allow=g729


 

 


SIP.CONF

 


[Dave]

; Cisco 7960 in Den, line 1

type=friend

username=Dave

password=qq

dtmfmode=rfc2833

host=dynamic

cancallforward=yes

qualify=yes

callerid=Dave in Den 560

context=cook-pbx

callgroup=1

pickupgroup=1

[EMAIL PROTECTED]

[201]

secret=

callerid=Jim Christie201

[EMAIL PROTECTED]

type=friend

host=dynamic

qualify=yes

port=5060

canreinvite=no

dtmfmode=rfc2833

context=cac

disallow=all

allow=ulaw

allow=g729

subscribecontext=blf_1

pickupgroup=1

callgroup=1


 

 


Extensions.conf

 


exten= 579,1,Dial(IAX2/SITEA_Server:[EMAIL PROTECTED]/menu)

exten=210,1,Dial(IAX2/SITEB_Server:[EMAIL PROTECTED]/menu)

 

 

Thanks for giving this the once over. I really hope I just am overlooking
something blatantly obvious.

- dbc.