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Richard Lyman | 1 Feb 2007 01:11

Re: how to get the status of failed call files

Rich Doughty wrote:
> i am creating call files, and catching successfully the ones that don't
> connect in a 'failed' extension. can anyone tell me how to find out the
> reason for the failure (ie busy, no answer).
>
> ${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't
> used) and channel_status doesn't seem to be any good.
>
> thanks in advance.
>
the event you received for OriginateFailure has a 'Reason: ' code.

that code breaks down as

0 = UNKNOWN FAILURE or DISCONNECT
3 = AST_CONTROL_RINGING (no answer)
5 = AST_CONTROL_BUSY
1 = AST_CONTROL_HANGUP
8 = AST_CONTROL_CONGESTION

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Shane Spencer | 1 Feb 2007 01:25
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Re: Re: [asterisk-dev] Dynamically Adding A Context

Hahaha,, I think thats a freaking SWEET suggestion :)

On 1/31/07, Andrew Furey <andrew.furey <at> gmail.com> wrote:
> On 01/02/07, Yuan LIU <yliu11 <at> hotmail.com> wrote:
> > What Lee suggested is to have the AGI script to actually parse, insert a new
> > context in extensions.conf, or deleting from it, then reload
> > extensions.conf.  This would at least achieve what you wanted to do.
>
> Or alternatively, to avoid complete disaster, why not have
> extensions.conf include another file (#include <somefile.conf>) and
> edit that one with your script? I've done that before (although I was
> actually recreating the entire file each time by populating from an
> external database).
>
> Andrew
>
> --
> Linux supports the notion of a command line or a shell for the same
> reason that only children read books with only pictures in them.
> Language, be it English or something else, is the only tool flexible
> enough to accomplish a sufficiently broad range of tasks.
>                           -- Bill Garrett
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Trevor Peirce | 1 Feb 2007 01:53
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Re: Toll-free dialing via PRI problem

Jerry Jones wrote:
> From asterisk, you do not hear anything other than ringing as it does 
> not cut the audio path through until it receives the answer from the 
> far end, hence the steady ringing.
So instead of Dial(Zap/g1/1800xxxxxxx,,r) just do 
Dial(Zap/g1/1800xxxxxxx,,) so early audio can make it through. Unless 
there's more to the puzzle?

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Jerry Jones | 1 Feb 2007 02:28

How would you compare feature set to a Metaswitch?

OK I need some help. Looking for comparisons for a large customer  
wishing to provide voip service over a region. We are up against  
Metaswitch who is claiming they can do anything Asterisk can do. I do  
not have too much information on Metaswitch so am looking for any  
information, preferably real world experience on how Asterisk and  
Metaswitch would compare side by side.

Thanks in advance.

Jerry
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Steve Prior | 1 Feb 2007 03:14

Which Java FastAGI implementation has the most "market share"?

When I was looking for a Java FastAGI interface for Asterisk I came 
across asterisk-java first and didn't realize there was more than one 
out there.  It seems to work fine and I've got my first project working 
with it, but I was wondering which Java FastAGI implementation is the 
most popular and how they compare against each other.

So I'm aware of:
asterisk-java
JastAGI
OrderlyCalls

Any comments on who the front runner is and why?

Steve
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Matthew Rubenstein | 1 Feb 2007 03:52

FreePBX/Debian Aborts Call While Connecting

	I used the "FreePBX on Debian" HowTo at
http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
to initiate calls to my SIP carrier. They get my registration, but they
see that my call is interrupted before they can complete the connection.
My Asterisk log shows that the call times out after the time (45s)
specified in my dialplan Dial() command. What is wrong?

[from /var/log/asterisk/full]:
Jan 30 23:40:35 DEBUG[6245] chan_sip.c: Stopping retransmission on
'24154c0d430e550821bda73c155cf573 <at> 82.165.187.196' of Request 102: Match
Found
Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command
'Command'
Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command
'Command'
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Scheduled a registration timeout
for 66.153.22.16 id  #17818 
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'7c2631854f167c817c1479d454825c1c <at> 82.165.187.196' Request 606: Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Stopping retransmission on
'7c2631854f167c817c1479d454825c1c <at> 82.165.187.196' of Request 606: Match
Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'7c2631854f167c817c1479d454825c1c <at> 82.165.187.196' Request 607: Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Stopping retransmission on
'7c2631854f167c817c1479d454825c1c <at> 82.165.187.196' of Request 607: Match
Found
Jan 30 23:40:44 DEBUG[6245] chan_sip.c: Registration successful
(Continue reading)

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Stephen Bosch | 1 Feb 2007 03:57
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kewlstart disconnect threshold

Hi, folks:

Can the loop drop detection threshold (normally defined in milliseconds)
be set on the Digium TDM-400 cards? Most PBXs let you set this value.

-Stephen-
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Tim Irvin | 1 Feb 2007 04:09
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Re: Toll-free dialing via PRI problem

Sheepishly, that was the magic bullet.  Thanks Trevor!!

Tim

Trevor Peirce <tpeirce <at> digitalcon.ca> wrote:
>
> Jerry Jones wrote:
>> From asterisk, you do not hear anything other than ringing as it does
>> not cut the audio path through until it receives the answer from the
>> far end, hence the steady ringing.
> So instead of Dial(Zap/g1/1800xxxxxxx,,r) just do
> Dial(Zap/g1/1800xxxxxxx,,) so early audio can make it through. Unless
> there's more to the puzzle? 

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Wayne Jensen | 1 Feb 2007 04:40

no lights on TE405P, but shows up in lspci, modules loaded

I pulled a working TE405P from one box and put it in another box.  I
compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no
lights on the card come on.

I do an lspci and the card shows up there.

I ran ztcfg -vv and got the error message Unable to open master device
'/dev/zap/ctl' so I followed the instructions in README.udev

the error message went away, but now when I run ztcfg I just get "0
channels configured"

Thanks!
Wayne
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Leo Ann Boon | 1 Feb 2007 04:54

Re: kewlstart disconnect threshold

Stephen Bosch wrote:
> Hi, folks:
>
> Can the loop drop detection threshold (normally defined in milliseconds)
> be set on the Digium TDM-400 cards? Most PBXs let you set this value.
>   
Good question. Anyone knows if the TDM-400 actually detect loop drops?

Leo

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Gmane