date:20060605

On Sun, 2006-06-04 at 02:02 -0500, Erick Perez wrote:
[snip]
 CFLAGS = -pipe -Wall -Wmissing-prototypes -Wmissing-declarations
 $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE
 #CFLAGS += -O2
 #CFLAGS += -O3 -march=pentium3 -msse -mfpmath=sse,387 -ffast-math
 # PERF: below is 10% faster than -O2 or -O3 alone.
 #CFLAGS += -O3 -ffast-math -funroll-loops
 # below is another 5% faster or so.
 CFLAGS += -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays
 -fsingle-precision-constant

Given that this is basically a Red Hat Enterprise Linux box I would
follow the way RH does their rpms:
- remove the -O3 or any other -O settings
- stick $(RPM_OPT_FLAGS) in the CFLAGS line which had the -O3 in it so
  rpm can decide which flags to use

Maybe you need to remove more -f flags. Don't know, my Makefile sourcery
is limited.

Regards,
Patrick

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Re: [Asterisk-Users] Microsoft CRM Asterisk

2006-06-05 Thread Arun Kumar

Hi Calvis,

Its good if I can help you in any why with this project.
thanks

../ArunOn 6/2/06, calvis [EMAIL PROTECTED] wrote:
Has anyone done any integration with Asterisk  Microsoft Dynamics CRM?Ijust wanted to check with the list before I pursue a project with the aboveintegration.In addition, if anyone would be interested in such an
integration let me know, and I will keep you posted on the results.Thanks,Charles AlvisInternet Technology Group, Inc.Redmond,WA___
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Re: [Asterisk-Users] transfer other features

On Sun, 2006-06-04 at 17:46 +0800, Ronald Wiplinger wrote:
 *CLI show features
 Builtin Feature   Default Current
 ---   --- ---
 Pickup*8  *8 
 Blind Transfer#   ## 
 Attended Transfer *2 
 One Touch Monitor *1 
 Disconnect Call   *   *0 

*0 is hardcoded in chan_zap and (iirc) is supposed to be a flash hook.
If you want to use *0 you have to change *0 in chan_zap.c to something
like x*0 and recompile. Do a search for *0 in chan_zap.c and you will
find it.

Regards,
Patrick

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Re: [Asterisk-Users] Meetme versus app_conference

On Sun, 2006-06-04 at 08:49 -0400, Matt Florell wrote:
 I kind of assume from all of the mentions of speex in the code that it
 is required.

Reading the Makefile posted by Erick Perez in an earlier posting I see:

# 0 = OFF 1 = astdsp 2 = speex
SILDET := 2

I guess if you specify 0 or 1 you don't need speex.

Regards,
Patrick


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[Asterisk-Users] Duplicate CDRs

2006-06-05 Thread Mark Drayton


Hi

For whatever reason we've getting 2 or 3 CDR lines logged for each call, 
often in different formats:

as1:~# grep test-89-1e2c /var/log/asterisk/cdr-csv/*.csv
/var/log/asterisk/cdr-csv/67.csv:67,88,89,test-context,88,SIP/test-88-2dae,SIP/test-89-1e2c,Dial,SIP/test-89|20,2006-06-05
 
11:41:31,,2006-06-05 11:41:35,4,0,NO ANSWER,DOCUMENTATION
/var/log/asterisk/cdr-csv/Master.csv:88,88,89,test-context,SIP/test-88-2dae,SIP/test-89-1e2c,Dial,SIP/test-89|20,2006-06-05
 
11:41:31,,2006-06-05 11:41:35,4,0,NO 
ANSWER,DOCUMENTATION,67,1149504091.85534,INT_CALL
/var/log/asterisk/cdr-csv/Master.csv:67,88,89,test-context,88,SIP/test-88-2dae,SIP/test-89-1e2c,Dial,SIP/test-89|20,2006-06-05
 
11:41:31,,2006-06-05 11:41:35,4,0,NO ANSWER,DOCUMENTATION

How can I configure asterisk not to log to accountcode.csv at all and 
only log the 18-field line (ie, with uniqueid and userfield) to 
Master.csv? I just want everything in one file, one line per record.

cdr.conf has only a '[general]' line -- nothing else.

cdr_custom.conf:

[mappings]
Master.csv = 
${CDR(clid)},${CDR(src)},${CDR(dst)},${CDR(dcontext)},${CDR(channel)},${CDR(dstchannel)},${CDR(lastapp)},${CDR(lastdata)},${CDR(start)},${CDR(answer)},${CDR(end)},${CDR(duration)},${CDR(billsec)},${CDR(disposition)},${CDR(amaflags)},${CDR(accountcode)},${CDR(uniqueid)},${CDR(userfield)}

cdr_manager.conf

[general]
enabled = yes

as1*CLI cdr status
CDR logging: enabled
CDR mode: simple
CDR registered backend: cdr-custom
CDR registered backend: cdr_manager
CDR registered backend: csv

Any ideas?

Thanks,

-- 

Mark Drayton
Frontier Systems
0207 420 4242


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Re: [Asterisk-Users] TDM-400 doesn't detect far-end hangup

2006-06-05 Thread Rich Adamson


Stephen Bosch wrote:

Hi:

I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with
kewlstart signalling.

When an outside caller calls the server, the TDM-400 goes off-hook and
provides a ringing tone to the caller. If the caller hangs up before the
receiving party answers the phone, Asterisk fails to detect the hang-up.
The TDM-400 stays off-hook, hogging the line, while Asterisk rings the
extension.

I thought the TDM-400 FXO module was supposed to detect far-end call drops.

Is there something I need to configure to make this work?


I'd first check to see if you are actually getting a disconnect 
indication from the pstn line. In North America, the disconnect can be 
seen with an ordinary voltmeter across tip-ring as a voltage drop (to 
zero) for about a 1/4 second. It usually occurs from 1 to 5 seconds 
after the pstn caller has hung up.


If you don't see that disconnect, then the issue is related to your pstn 
provider not providing it to you.


If you do see that disconnect, then you've got something misconfigured 
in asterisk.



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Re: [Asterisk-Users] Campusing two Asterisk boxes?

2006-06-05 Thread Sean Cook

Yes... it is very easy to do...

; on box a
exten = _NXXNXX,1,DIal(IAX2/boxb/${EXTEN})


;on box b
exten = _NXXNXX,1,Dial(IAX2/boxa/${EXTEN})


you just need to make sure that the context on the each side will have a
match for passing in ${EXTEN} to the other side

[from-boxa]
exten = _NXXNXX,1,Dial(ZAP/g0/${EXTEN})


[from-boxb]
exten = _NXXNXX,1,Dial(ZAP/g0/${EXTEN})


 I have been looking around some and I can't seem to find anything which
 will
 answer my question.  If I have two Asterisk boxes in different locations
 which
 are linked to each other over the internet, can I configure the boxes to
 use
 each other's lines as local?

 In other words, let's say Site A has Phone1
 for a 1FB line going into it on an FXO port.  Site B has Phone2 for a 1FB
 line going into it on an FXO port.  Is there a way to configure Site A to
 use Phone2 from Site B and vice versa?

 Undrhil
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Re: [Asterisk-Users] WCTDM-24xxp woes

2006-06-05 Thread Rich Adamson


Andrew D Kirch wrote:

I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N
Nforce based board).  I am using a Digium wctdm24xxp to terminate 8 (2x
quad) FXO lines.  Using ztmonitor (From zapata) I cannot see that there is
any registerable incoming volume from these lines.  I've been running them
at rxgain = 25 (zapata.conf) to make the audio audible, however this
creates poor call quality issues (static and distortion) on most calls,
and audio garble in voicemails.   Fxotune fails for every line with Could
not fill input buffer
I've tried changing PCI slots, played with echo settings, and done
everything else I can think of to make this card play nice to no avial.

Anyone with solutions or ideas, your input will be greatfully appreciated.


As others have already noted, start with rxgain  txgain set to 0. And, 
check the lines with an analog phone to ensure you don't have weak audio 
from the pstn line to start with.


If you do have weak audio (as in a very long pstn line), you're not 
likely to get the TDM400 or TDM2400 card to compensate for the loss 
without seriously impacting the s/w echo canceler.


For high loss pstn lines, ztmonitor will not provide any useful indication.

If you approach the problem from a professional perspective, you would 
use a transmission test set to measure the loss of the pstn line by 
dialing into the central office milliwatt generator (no asterisk 
involvement). If that measured pstn loss is anything greater then about 
7db to 10db, I'd contact your pstn provider to see if there is anything 
they can do to improve it. (Most US telco's can install repeaters in the 
central office that will boast the audio levels. Repeaters have been in 
use by telco's for at least 20 years, and are typically used on long 
rural pstn lines.)


If the analog audio is reasonable (or your measured pstn loss is less 
then about 7db), then you've got something wrong with the TDM2400 
installation. That could be something like a mis-wired TDM2400-to-pstn 
line connections, etc. It will have nothing at all to do with the pci 
bus, interrupts, etc.



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[Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-05 Thread Esteban Guana-Jarrin


Thanks Armin


The call is rejected by Asterisk, so it looks like your dialplan
has no rule for accepting calls to '99546476'.



Armin



My dial plan as shown below is,

[capi-in]
exten = s,1,Dial(Sip/123,20)
exten = s,2,Voicemail(123)
exten = s,3,Hangup

I believe I should be able to receive calls with the above.

I have also tried the following, and i get the same problem and debug output 
is the same.


[capi-in]
exten = 99546476,1,Dial(Sip/123,20)
exten = 99546476,2,Voicemail(123)
exten = 99546476,3,Hangup

Any other ideas ???

Esteban

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Re: [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-05 Thread Armin Schindler

On Mon, 5 Jun 2006, Esteban Guana-Jarrin wrote:
 Thanks Armin
 
   The call is rejected by Asterisk, so it looks like your dialplan
   has no rule for accepting calls to '99546476'.
 
   Armin
 
 
 My dial plan as shown below is,
 
 [capi-in]
 exten = s,1,Dial(Sip/123,20)
 exten = s,2,Voicemail(123)
 exten = s,3,Hangup
 
 I believe I should be able to receive calls with the above.

No, 's' is not used. The called number must be used.
 
 I have also tried the following, and i get the same problem and debug output
 is the same.
 
 [capi-in]
 exten = 99546476,1,Dial(Sip/123,20)
 exten = 99546476,2,Voicemail(123)
 exten = 99546476,3,Hangup
 
 Any other ideas ???

That looks correct. Is the context= in capi.conf really set to capi-in ?

Armin

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RE: [Asterisk-Users] chan_capi-cm-0.6 and incoming calls problem

2006-06-05 Thread James Harper

 My dial plan as shown below is,
 
 [capi-in]
 exten = s,1,Dial(Sip/123,20)
 exten = s,2,Voicemail(123)
 exten = s,3,Hangup
 
 I believe I should be able to receive calls with the above.

With immediate = yes then you should.

 I have also tried the following, and i get the same problem and debug
 output is the same.
 
 [capi-in]
 exten = 99546476,1,Dial(Sip/123,20)
 exten = 99546476,2,Voicemail(123)
 exten = 99546476,3,Hangup
 
 Any other ideas ???

Turn on asterisk debugging too. Capi seems to be working okay, maybe
asterisk isn't picking up the call for some reason. Maybe: 

asterisk -r
set verbose 9
set debug 9
capi debug

then make an incoming call and copy the output into an email and send it
to the list (unless it is really really long, then you may have to look
for interesting bits).

You should see a message in there somewhere that tells you that either
the capi driver is rejecting the call because it doesn't want to answer
that msn (your earlier logs make that unlikelye), or that asterisk can't
find an extension for it.

James
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Re: [Asterisk-Users] fine-tuning asterisk questions

2006-06-05 Thread William Piper

My apologies, I didn't realize I was speaking to someone else.
As far as I know the dialplan does not need to have the j option to do N+101. I'm using 1.2.7.1 without the j option and it jumps fine.

I suppose that will work fine as long as you turn off call waiting on the phone itself. I provision everything from the server side so I don't have to provision the boxes. Unless you use the same SIP phones for everyone, it could be a pain to do on the phone side... and what happens if you want to give1 personthe ability to do call waiting without giving it to everyone? Even if you did the cfg files from a tftp, you'd still have to get the MAC address provision a new cfg file what a pain!

I say, depending on the size of your company, create a dialplan with a star feature to activate deactivate call waiting. Just do a 'DBput and dump the calleridnum value in the database, then do a DBget on your incoming dialplan to see if call waiting is activated for that user. That is super simple and leaves it up to the end user, or if you want... don't write the star feature just provision it from the DB. That way it leaves it up to you and the end user still can't change it.

bp
On 6/5/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote: Yes you are correct... by default asterisk will send the call to priority
N+101... what is your point? You asked about turning off call waiting.In the example that I provided, if the amount of active calls is 1 then
it will forward to VM without dialing the exten. That is what you asked for... right? bp Nope.I am a different poster just wanting to clarify (for myself) that Asterisk would do exactly what the original poster
wanted without any special programming.I wasn't aware that there would be any kind of notification to the station being called that there was a second call incoming.Everything I've read so far just says that if the station
is in use, the call is routed to priority n + 101 as a busy call.Only if you use the j option in the dial command.In previous versionsit did it automatically:'j' -- Jump to n+101 if all of the channels were busy.
--Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)___
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Re: [Asterisk-Users] Busy Signals after hangup

On Saturday 03 June 2006 14:06, Rick Smith wrote:
 The call gets made, I leave a voicemail, or complete the call in some
 manner, and the other side hangs up.  I hear a busy signal on the phone
 on my end.

I'll bet a donut it's not a busy signal but rather a fast busy which is 
known as a congestion signal.

Asterisk is giving that to you because it has nothing else to do in the 
dialplan.  This is what I do on all my installations to make it behave more 
like the phone company:

exten = 199,1,Answer()
exten = 199,n,Dial(SIP/100,20,g)
exten = 199,n,Macro(handle-hangup)

(the trick is to make sure that 'g' is in the Dial() options, as it instructs 
Dial() to continue on in the dialplan after the channel is hung up.  An 
alternative would be to trap the 'h' (hangup) extension and call the macro as 
well.

The macro is pretty straightforward.  Don't be put off by its size.  It tries 
to do the right thing, and can handle PRI hangup causes:

---8--8--8---
; handle hangup macro
; this macro attempts to go though and do something intelligent with the 
HANGUPCAUSE and DIALSTATUS
[macro-handle-hangup]
exten = s,1,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is 
${DIALSTATUS})
exten = s,n,GotoIf($[${HANGUPCAUSE} = 0]?s,nohc)
exten = s,n,Goto(hc-${HANGUPCAUSE},1)
exten = s,n(nohc),GotoIf($[${DIALSTATUS} = ANSWER]?hc-16,1)
exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?hc-17,1)
exten = s,n,GotoIf($[${DIALSTATUS} = NOANSWER]?hc-19,1)
exten = s,n,GotoIf($[${DIALSTATUS} = CONGESTION]?hc-42,1)
exten = s,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?hc-44,1)
exten = s,n,GotoIf($[${DIALSTATUS} = CANCEL]?hc-0,1)
exten = s,n,Goto(hc-0,n)

exten = hc-0,1,NoOp(${HANGUPCAUSE} - Not Defined)
exten = hc-0,n,Goto(ind-congestion,1)
exten = hc-1,1,NoOp(${HANGUPCAUSE} - Unallocated)
exten = hc-1,n,Goto(ind-outofservice,1)
exten = hc-2,1,NoOp(${HANGUPCAUSE} - No Route to Transit Network)
exten = hc-2,n,Goto(ind-congestion,1)
exten = hc-3,1,NoOp(${HANGUPCAUSE} - No Route to Destination)
exten = hc-3,n,Goto(ind-congestion,1)
exten = hc-6,1,NoOp(${HANGUPCAUSE} - Channel Unacceptable)
exten = hc-6,n,Goto(ind-congestion,1)
exten = hc-7,1,NoOp(${HANGUPCAUSE} - Call Awarded Delivered)
exten = hc-7,n,Goto(ind-hangup,1)
exten = hc-16,1,NoOp(${HANGUPCAUSE} - Normal Clearing)
exten = hc-16,n,Goto(ind-hangup,1)
exten = hc-17,1,NoOp(${HANGUPCAUSE} - User Busy)
exten = hc-17,n,Goto(ind-busy,1)
exten = hc-18,1,NoOp(${HANGUPCAUSE} - No User Response)
exten = hc-18,n,Goto(ind-hangup,1)
exten = hc-19,1,NoOp(${HANGUPCAUSE} - No Answer)
exten = hc-19,n,Goto(ind-hangup,1)
exten = hc-21,1,NoOp(${HANGUPCAUSE} - Call Rejected)
exten = hc-21,n,Goto(ind-outofservice,1)
exten = hc-22,1,NoOp(${HANGUPCAUSE} - Number Changed)
exten = hc-22,n,Goto(ind-outofservice,1)
exten = hc-27,1,NoOp(${HANGUPCAUSE} - Destination Out-of-Order)
exten = hc-27,n,Goto(ind-outofservice,1)
exten = hc-28,1,NoOp(${HANGUPCAUSE} - Invalid Number Format)
exten = hc-28,n,Goto(ind-congestion,1)
exten = hc-29,1,NoOp(${HANGUPCAUSE} - Facility Rejected)
exten = hc-29,n,Goto(ind-congestion,1)
exten = hc-30,1,NoOp(${HANGUPCAUSE} - Response to Status Enquiry)
exten = hc-30,n,Goto(ind-hangup,1)
exten = hc-31,1,NoOp(${HANGUPCAUSE} - Normal Unspecified)
exten = hc-31,n,Goto(ind-hangup,1)
exten = hc-34,1,NoOp(${HANGUPCAUSE} - Normal Circuit Congestion)
exten = hc-34,n,Goto(ind-congestion,1)
exten = hc-38,1,NoOp(${HANGUPCAUSE} - Network Out-of-Order)
exten = hc-38,n,Goto(ind-congestion,1)
exten = hc-41,1,NoOp(${HANGUPCAUSE} - Normal Temporary Failure)
exten = hc-41,n,Goto(ind-congestion,1)
exten = hc-42,1,NoOp(${HANGUPCAUSE} - Switch Congestion)
exten = hc-42,n,Goto(ind-congestion,1)
exten = hc-43,1,NoOp(${HANGUPCAUSE} - Access Information Discarded)
exten = hc-43,n,Goto(ind-hangup,1)
exten = hc-44,1,NoOp(${HANGUPCAUSE} - Requested Channel Unavailable)
exten = hc-44,n,Goto(ind-congestion,1)
exten = hc-45,1,NoOp(${HANGUPCAUSE} - Pre-Empted)
exten = hc-45,n,Goto(ind-congestion,1)
exten = hc-50,1,NoOp(${HANGUPCAUSE} - Facility Not Subscribed)
exten = hc-50,n,Goto(ind-congestion,1)
exten = hc-52,1,NoOp(${HANGUPCAUSE} - Outgoing Call Barred)
exten = hc-52,n,Goto(ind-congestion,1)
exten = hc-54,1,NoOp(${HANGUPCAUSE} - Incoming Call Barred)
exten = hc-54,n,Goto(ind-congestion,1)
exten = hc-57,1,NoOp(${HANGUPCAUSE} - Bearer Capability Not Authorized)
exten = hc-57,n,Goto(ind-congestion,1)
exten = hc-58,1,NoOp(${HANGUPCAUSE} - Bearer Capability Not Available)
exten = hc-58,n,Goto(ind-congestion,1)
exten = hc-65,1,NoOp(${HANGUPCAUSE} - Bearer Capability Not Implemented)
exten = hc-65,n,Goto(ind-congestion,1)
exten = hc-66,1,NoOp(${HANGUPCAUSE} - Channel Not Implemented)
exten = hc-66,n,Goto(ind-congestion,1)
exten = hc-69,1,NoOp(${HANGUPCAUSE} - Facility Not Implemented)
exten = hc-69,n,Goto(ind-congestion,1)
exten = hc-81,1,NoOp(${HANGUPCAUSE} - Invalid Call Reference)
exten = hc-81,n,Goto(ind-congestion,1)
exten =

Re: [Asterisk-Users] Inconsistency with ANI and channel callerid


- Gil Kloepfer [EMAIL PROTECTED] wrote:

 It would seem that the right behavior would be one of consistency --
 if
 someone specifies the callerid= option in any of the channel .conf
 files,
 then it should either set or not set ANI, but not behave differently
 for
 different channels.

Agreed. If different channel drivers are setting the CLID/ANI differently when 
the CLID it set via their configuration systems, then this is a bug. Please 
report it on Mantis so we can track it and get someone to correct it.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Config Revision Control

On Saturday 03 June 2006 02:47, Michiel van Baak wrote:
 I use subversion for this. Every server has its own branch.
 There's also a branch called 'common'
 All the server specific branches are svn-copied and svnmerge
 init from this branche.
 Then the svn automerge thingie Kevin wrote for the asterisk
 svn tree is automerging changes to the 'common' tree to all
 the server trees.
 In the server trees I make changes specific for one server.

Can you give some more details?  I am VERY interested in this!

-A.
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Re: [Asterisk-Users] reinvite


- Osama Kamal [EMAIL PROTECTED] wrote:
 I am running asterisk behind nat, and 2 sip phones on 2 different adsl
 neted connections, asterisk is staying always in rtp media path, while
 canreinvite=yes is configured in both extensions. I need asterisk to
 stay away from the rtp media path, what is wrong with that setup?

It is nearly impossible to get a direct media path between two endpoints that 
are both behind NATs, regardless of the SIP server/proxy you use. Asterisk is 
no different in this regard.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Asterisk and SATA Raid 1


- mustardman29 [EMAIL PROTECTED] wrote:

 I know that Digium and FreePBX were not recommending it awhile back
 but I
 think that was based on 2.4 Kernel and Digium hardware issues.  I am

Can you give me a pointer to any place where Digium recommended against using 
hardware RAID cards? I can't imagine that being an issue for Asterisk or any of 
our hardware.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Prices of g729 codec

On Saturday 03 June 2006 16:57, trixter aka Bret McDanel wrote:
 but $10 only gets you one license, what if you are vonage sized and need
 to support a million customers?  What if you accept that you can settle

If you are Vonage and need to support a million customers I will bet you are 
not transcoding a million conversations on regular PC hardware.  You will 
have AS5300/5400 boxes or MaxTNTs, which you have already paid WELL MORE than 
$2M for, which INCLUDES the AudioCodes patent license for g.729.

You can't avoid it and stay legit.  Sorry to burst your bubble.

 for a 5:1 ratio, then its only 200,000 or $2M.  Just for codec licenses,
 not to mention all the other costs of being a business.  What if you are
 smaller than vonage, say 10k channels in use, then that smaller entity,
 probably without the hundreds of millions of VC that vonage got you
 would have to come up with $100k.  Still more than $10.

Again, 10k channels you'll have a half dozen MaxTNT boxes terminating DS3s.  
Your fixed costs will already be significantly higher and that little $10 
license fee is included in that.

 If you are going to bring businesses into it, at least accept that a
 business would most likely pay more than $10 for their licensing needs.

Nonsense.  The license fee that Digium charges is for onesie-twosie stuff.  If 
you're making a real go of this as a business you will be paying that patent 
license fee either through Digium (if you're transcoding on PCs, in which 
case you are either doing something different or you're just plain stupid) or 
you are paying a smaller patent license fee which was included in the price 
of the hardware on your NAS equipment.

 no its not that they want quantity becuase they will sell just one
 license, they only want to deal with people that implement the systems
 not the end users of the system.  They claim the reasoning for this is
 to make it easier for end users to know that they have licenses -
 basically if you have it you are licensed.  Even if that isnt the case.
 Check www.sipro.com for more info on g729 licensing.

It's always easier to work with businesses and deal with quantity than it is 
to deal with the public or end customer and all the hassles with that.  I'm 
positive that AudioCodes doesn't want to staff a customer service department 
to deal with Joe Sixpack who's cousin's friend's son is a computer whiz and 
hooked up this phone over teh intarweeb thingie for him but he just can't 
it working perfect.

-A.
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Re: [Asterisk-Users] Duplicate CDRs


- Mark Drayton [EMAIL PROTECTED] wrote:

 How can I configure asterisk not to log to accountcode.csv at all
 and 
 only log the 18-field line (ie, with uniqueid and userfield) to 
 Master.csv? I just want everything in one file, one line per record.

You have both cdr_csv and cdr_custom loaded, with cdr_custom configured to 
write into Master.csv. Asterisk is doing exactly what you told it to do... 
write the CDR twice into Master.csv.

If you only want one of them, load the module for the one you want and don't 
load the other one.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Busy Signals after hangup

2006-06-05 Thread Aaron Daniel


On Mon, 5 Jun 2006, Andrew Kohlsmith wrote:

I'll bet a donut it's not a busy signal but rather a fast busy which is
known as a congestion signal.


I'll be a jelly filled donut that it's the device he's using and not 
asterisk sending the signal :)  We have a few ATA's that don't 
automatically hang up even though the call has ended, they just do the 
congestion symbol.  It's caught me off guard a couple times.



--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] fine-tuning asterisk questions


- William Piper [EMAIL PROTECTED] wrote:

 My apologies, I didn't realize I was speaking to someone else.
 As far as I know the dialplan does not need to have the j option to
 do N+101. I'm using 1.2.7.1 without the j option and it jumps fine.

This is true in Asterisk 1.2.x, as the default in the code is to enable jumping 
(but the default in the sample extensions.conf file is to have jumping turned 
off). In Asterisk 1.4 the default in the code will be to have jumping disabled, 
and it will need to be turned on globally (or on an application basis) to use 
be used. In Asterisk 1.6 it will be gone forever :-)

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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[Asterisk-Users] collect call

2006-06-05 Thread Virmones Pereira Tavares de Miranda









I
need help with collet call

This my case:

Line PSTN -- ASTERISK MFC/R2 -- PBX

when receive call in the PSTN the asterisk send this call to PBX, but if PBX
its
enabled block collect call, the ASTERISK hang up call;

this block call its not category 8 or 9, it is make with polary reverse

its possible solve this problem?






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Re: [Asterisk-Users] Prices of g729 codec

On Saturday 03 June 2006 04:05, Sahil Gupta wrote:
 We recently had around 60-80 licenses become useless because Digium
 refused to renew the keys on that.  That was a bit of money kissed
 goodbye.

Ok, that's a great fairy tale.  Now tell us the true story.

When you buy licenses from Digium, you register them and they are branded 
with information from your machine (most likely MAC address of the NIC, but 
I'm not 100% sure nor do I particularly care for the details.)

If you upgrade hardware you may have to re-register them.  Digium allows you 
to automatically re-register once without phone calls or any explanation.  
After that, you cannot re-register without calling Digium and making a case 
for it.  This is a restriction placed on them by the patent holders of the 
g.729 codec.

So the TRUE story is that you had 60-80 licenses, registered, changed your 
hardware, re-registered, changed your hardware AGAIN and for one reason or 
another failed to convince Digium that it was a legitimate change to warrant 
a re-registration.

I have personally called Digium and provided sufficient reasoning to grant me 
a third registration.  

So honestly now, Sahil, what did you guys do that was so different?  It 
*really* pisses me off when people like you give a half-assed, half-baked 
digium sucks post.  If you've got an honest beef with Digium, then sure, 
lay it all out, but don't present half the fucking story and then bitch about 
how the big bad Digium beat you up and stole your lunch money.

... just like my kids...  Wh, Joshua hit me!  Yeah but you've been 
bugging the shit out of him for the last 5 minutes and he asked you nicely to 
stop twice.  I'd have hit you too, Katie.

-A.
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[Asterisk-Users] RE: Size limitations of extensions.conf

2006-06-05 Thread Brent Torrenga

If you need to do a couple differing operations on a list of many
area/country codes, then you may consider using the database to let the dial
plan choose what to do, rather than go through so many extensions.

I mention this to keep your extensions.conf easier to read, not because I
know whether or not a long extensions.conf will break things...

 Can someone tell me the size (or any other) limitations for the
extensions.conf?
 
 We have managed to keep our file pretty small thanks to AGI but we are
 about to setup a bunch of call restrictions based on area and country
 code.
 
 One line per area code in the US alone adds a LOT of text to this file.
 
 Is it a bad thing to have 5 or 6000 lines of text in your
 extensions.conf on a production system?
 
 Will it affect the performance?


Sincerely,

Brent A. Torrenga

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:[EMAIL PROTECTED]
web:www.torrenga.com

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Re: [Asterisk-Users] reinvite

2006-06-05 Thread Osama Kamal

does thia apply on SIP only or also IAX?On 6/5/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Osama Kamal [EMAIL PROTECTED] wrote:
 I am running asterisk behind nat, and 2 sip phones on 2 different adsl neted connections, asterisk is staying always in rtp media path, while canreinvite=yes is configured in both extensions. I need asterisk to
 stay away from the rtp media path, what is wrong with that setup?It
is nearly impossible to get a direct media path between two endpoints
that are both behind NATs, regardless of the SIP server/proxy you use.
Asterisk is no different in this regard.--Kevin P. FlemingSenior Software EngineerDigium, Inc.___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Allowing multiple exchanges

2006-06-05 Thread Kevin Smith


Hey Doug,

Few things you can do. First off, are the numbers for incoming callers 
or for when you are making a call? One way that we do it because our 
numbers change a lot is I have a text file with all the numbers on it. 
Like below:


[localtoolexchange]
exten = _342, 1, Goto(whereever)
etc..

and then I include them where you need them. Now this is for outgoing, 
for incoming you just would need to remove the _. Now if it is a range 
of numbers that you know you can do the following:


exten = _[12347-9][2-6789]X, Goto(whereever)

The first part will look for 1,2,3,4,7,8, and 9. The second 
2,3,4,5,6,7,8,9, and finally X is 0-9.


If you have them in a database, I would use the text file method. It is 
easy to write a script to build a new file and reload it into asterisk. 
But you also can write the second part of the script with a little more 
tinkering.


Kevin


Doug Crompton wrote:

What is the best way to include a whole group of exchanges into a dial
plan? I want to route local toll free by exchange (first three) and I will
have a bunch. Can they be stored somewhere and compared as a group to that
position in the dialplan?

Doug

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Re: [Asterisk-Users] Config Revision Control

2006-06-05 Thread Michiel van Baak

On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote:
 On Saturday 03 June 2006 02:47, Michiel van Baak wrote:
  I use subversion for this. Every server has its own branch.
  There's also a branch called 'common'
  All the server specific branches are svn-copied and svnmerge
  init from this branche.
  Then the svn automerge thingie Kevin wrote for the asterisk
  svn tree is automerging changes to the 'common' tree to all
  the server trees.
  In the server trees I make changes specific for one server.
 
 Can you give some more details?  I am VERY interested in this!

Most is already in my previous mail.

This is my layout:
branches/common
branches/servers/home001
branches/servers/home002
branches/servers/cust001

Like that, you get the idea
The branches/common holds a full config, cept for sip users etc. So
all the [global] and [default] stuff. Also the
extensions.conf has some macro's and contexts I need on
every machine.

The home001 etc hold the conf I actually run on a server.
All the specific sip and iax peers/users are defined in it.
Also the specific stuff for extensions.conf for that server.

If I for example want the congestion in my default outbound
routing macro to play congestion for 5 seconds instead of 10
I only alter extensions.conf in branches/common
The automerge will take care of the promoting it to all the
other branches.

I use this script to do the automerging every hour:
http://svn.digium.com/view/repotools/svn-automerge?rev=54view=markup
This also means you have to use the modified svnmerge from
the asterisk project:
http://svn.digium.com/view/repotools/svnmerge?rev=63view=markup

All my servers do auto svn up of the asterisk configs.

I hope this is enough details...
-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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Re: [Asterisk-Users] fine-tuning asterisk questions

2006-06-05 Thread William Piper


On 6/5/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: 
This is true in Asterisk 1.2.x, as the default in the code is to enable jumping (but the default in the sample 
extensions.conf file is to have jumping turned off). In Asterisk 1.4 the default in the code will be to have jumping disabled, and it will need to be turned on globally (or on an application basis) to use be used. In Asterisk 
1.6 it will be gone forever :-)


By gone forever in 1.6... do you mean that even the j in the dial plan won't work either? Will it just go to the next priority in the event of a congested or busy signal? 

I assume goto will still work... right?
bp
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Re: [Asterisk-Users] Duplicate CDRs

2006-06-05 Thread Mark Drayton







Kevin P. Fleming
[EMAIL PROTECTED] 
Sent by: [EMAIL PROTECTED]
05/06/2006 14:48



Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com





To
Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com


cc



Subject
Re: [Asterisk-Users] Duplicate CDRs









- Mark Drayton [EMAIL PROTECTED] wrote:

  How can I configure asterisk not to log to accountcode.csv
at all
  and 
  only log the 18-field line (ie, with uniqueid and userfield)
to 
  Master.csv? I just want everything in one file, one line per
record.

 You have both cdr_csv and cdr_custom loaded, with cdr_custom configured
to write into Master.csv. Asterisk is doing exactly what you told 
it to do... write the CDR twice into Master.csv.

 If you only want one of them, load the module for the one you want
and don't load the other one.

Okay. Which one writes the 18-field line (with uniqueid
and userfield)?cdr_custom.conf has fields for these two but the wiki docs
also say that cdr_csv will write uniqueid and userfield if configured.
Can I just unload whichever one I don't need to stop it writing or do I
need to reload the logging/cdr system? Presumably cdr_csv that writes accountcode.csv
as cdr_custom specifies Master.csv. 

Can I set enabled=no in cdr_manager.conf if I just
want plain CDRs? Can I unload that module?

Sorry for the questions; there's not a great deal
of documentation on how this hangs together.

Thanks,

-- 

Mark Drayton
Frontier Systems
0207 420 4242


This message and any attachment are confidential and may be privileged or
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[Asterisk-Users] Can´t send emails

Hello everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly, so I can receive them directly from my phone with *97, but still the Asterisk don´t send it through Internet.As I have seen the SMTP is working, and I can ping any site of Internet, and any of my internal addresses, but still I have no idea about what is happening.Thanks.Yrving __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/ ___
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Re: [Asterisk-Users] TDM PCI Master Abort

2006-06-05 Thread John Novack




Stephen Bosch wrote:


Jeremy McNamara wrote:
 


Stephen Bosch wrote:

   


I don't know. I'll have to check. Is that a requirement?
 



Yes - Most absolutely.


http://www.digium.com/en/products/hardware/tdm400p.php
   



I've confirmed that the board supports PCI 2.2.

I've also updated the BIOS on the motherboard, but the problem is still
there.

I'm going to try moving the card to a different PCI slot (a desperation
move).

-Stephen-
 


For what it's worth -
I had a similar problem with a MB that was 2.2, but the card was not seen.
Digiums response was try another motherboard

The TDM 400 does not work in all PCI 2.2 MB configurations.
The Sangoma A200 worked, and is still working, in that same MB.

John Novack

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Re: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Matt Riddell (IT)

 So honestly now, Sahil, what did you guys do that was so different?  It 
 *really* pisses me off when people like you give a half-assed, half-baked 
 digium sucks post.  If you've got an honest beef with Digium, then sure, 
 lay it all out, but don't present half the fucking story and then bitch about 
 how the big bad Digium beat you up and stole your lunch money.
 
 ... just like my kids...  Wh, Joshua hit me!  Yeah but you've been 
 bugging the shit out of him for the last 5 minutes and he asked you nicely to 
 stop twice.  I'd have hit you too, Katie.

ROFL!!!

:) So eloquently put!

-- 
Cheers,

Matt Riddell
___

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http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
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Re: [Asterisk-Users] Asterisk and SATA Raid 1

2006-06-05 Thread Julian J. M.


Hi,

I also remember reading that.. but i'm not sure if it was Digium's
word ;) It had to do with some SCSI and SATA controllers taking
control of the PCI bus for too much time, and causing frame-slips or
IRQ losses on TDM hardware.

Julian.

On 6/5/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:


- mustardman29 [EMAIL PROTECTED] wrote:

 I know that Digium and FreePBX were not recommending it awhile back
 but I
 think that was based on 2.4 Kernel and Digium hardware issues.  I am

Can you give me a pointer to any place where Digium recommended against using 
hardware RAID cards? I can't imagine that being an issue for Asterisk or any of 
our hardware.

--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Paul

Andrew Kohlsmith wrote:

Nonsense.  The license fee that Digium charges is for onesie-twosie stuff.  If 
you're making a real go of this as a business you will be paying that patent 
license fee either through Digium (if you're transcoding on PCs, in which 
case you are either doing something different or you're just plain stupid) or 
you are paying a smaller patent license fee which was included in the price 
of the hardware on your NAS equipment.
  

I really doubt that Digium would insist on the $10 fee for a quantity buyer.


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Re: [Asterisk-Users] Can´t send emails

2006-06-05 Thread Alex Robar

Yrving,How have you verified that your SMTP is working? Can you actually send a message from your Asterisk server to one of your internal addresses?AlexOn 6/5/06, 
yrving rivas [EMAIL PROTECTED] wrote:
Hello everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly, so I can receive them directly from my phone with *97, but still the Asterisk don´t send it through Internet.
As I have seen the SMTP is working, and I can ping any site of Internet, and any of my internal addresses, but still I have no idea about what is happening.Thanks.Yrving
 __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - 
http://correo.yahoo.com.mx/ 
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED]
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Re: [Asterisk-Users] Can´t send emails

2006-06-05 Thread David K Parker

Many ISPs block outbound SMTP except directly to their mail servers. If this is the case, you could try using a mailhop service such as one provided by dnydns.org.
On 6/5/06, yrving rivas [EMAIL PROTECTED] wrote:
Hello everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly, so I can receive them directly from my phone with *97, but still the Asterisk don´t send it through Internet.
As I have seen the SMTP is working, and I can ping any site of Internet, and any of my internal addresses, but still I have no idea about what is happening.Thanks.Yrving
 __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - 
http://correo.yahoo.com.mx/ 
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Re: [Asterisk-Users] New Member, saying Hi. :)

2006-06-05 Thread Lewis Agosta

Welcome to our world. You will find yourself up nights thinking of all the possibilities and kicking yourself all day because you can't get them all done :)

Good luck.
On 4 Jun 2006 06:02:39 -, [EMAIL PROTECTED] 
[EMAIL PROTECTED] wrote:
Hello everyone.I had heard about this open-source PBX once a while back.I wasn't too interested in it at the time but I kept the info filed away
for possible future use.A couple of days ago, I was walking around Barnesand Nobles and I found this book, called Asterisk: The Future of Telephony.I paged through it a little and I was really excited by what I read.Then
I remembered the open-source PBX I had read about before: it was Asterisk!This book was about that open-source PBX.It was very enlightening and Idecided to buy the book so I could learn more.When I got home, I read
through a few chapters and I also started looking online to find a download.I somehow managed to find a ready-made appliance called PoundKey whichI downloaded and installed on my spare PC.Now I got confused because I wasn't
sure where to go from the command-line prompt.So, I'm starting over at squareone and I am going to download plain-jane Asterisk and get it running on aKnoppix HD installation... I hope.:)Anyway, this has been a brief (trust
me, brief is good!) introduction of myself to the group.I'm sure I'll beasking lots of questions.:)Undrhil___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Prices of g729 codec

On Mon, 2006-06-05 at 09:49 -0400, Andrew Kohlsmith wrote:
 On Saturday 03 June 2006 16:57, trixter aka Bret McDanel wrote:
  but $10 only gets you one license, what if you are vonage sized and need
  to support a million customers?  What if you accept that you can settle
 
 If you are Vonage and need to support a million customers I will bet you are 
 not transcoding a million conversations on regular PC hardware.  You will 
 have AS5300/5400 boxes or MaxTNTs, which you have already paid WELL MORE than 
 $2M for, which INCLUDES the AudioCodes patent license for g.729.
 
 You can't avoid it and stay legit.  Sorry to burst your bubble.
 

my bubble wasnt bursted as you proved my point. Thank you for proving
that for me so I didnt have to.  I was responding to the comments that
said it was only $10 for a license, and not any comments on larger
entities.  


  for a 5:1 ratio, then its only 200,000 or $2M.  Just for codec licenses,
  not to mention all the other costs of being a business.  What if you are
  smaller than vonage, say 10k channels in use, then that smaller entity,
  probably without the hundreds of millions of VC that vonage got you
  would have to come up with $100k.  Still more than $10.
 
 Again, 10k channels you'll have a half dozen MaxTNT boxes terminating DS3s.  
 Your fixed costs will already be significantly higher and that little $10 
 license fee is included in that.
 
Its not $10, which also goes along with something else I mentioned
elsewhere.  Digium charges $10 but the max cost for a g729 license is
about $1.25.  It goes down to about $0.10/license in quantity.  As such
it doesnt add a whole lot to the cost of the device once the initial
code is in place (as that development does have cost since the license
fee doesnt cover any implementation, only the right to sell that
implementation).

And again to clarify, since this was aparently lost somewhere, I was
responding to the mentality that everyone is a home user and its only
$10 for a license and that is all anyone ever needs to pay.  You have
proven me right, thanks again for that.  When you get out of the home
user mindset the cost goes up dramatically and the argument that I
responded to that the cost isnt that high at $10/license was invalid,
even though you seem to be saying that it is that same cost, which
anyone who really knows anything about the licensing knows that isnt
true.


  If you are going to bring businesses into it, at least accept that a
  business would most likely pay more than $10 for their licensing needs.
 
 Nonsense.  The license fee that Digium charges is for onesie-twosie stuff.  
 If 
 you're making a real go of this as a business you will be paying that patent 
 license fee either through Digium (if you're transcoding on PCs, in which 
 case you are either doing something different or you're just plain stupid) or 
 you are paying a smaller patent license fee which was included in the price 
 of the hardware on your NAS equipment.
 

My point again, thanks for the recap.  The $10/license fee outside of
the home user market is what I was contesting.  Why do you keep proving
my point in such an argumentative way?


  no its not that they want quantity becuase they will sell just one
  license, they only want to deal with people that implement the systems
  not the end users of the system.  They claim the reasoning for this is
  to make it easier for end users to know that they have licenses -
  basically if you have it you are licensed.  Even if that isnt the case.
  Check www.sipro.com for more info on g729 licensing.
 
 It's always easier to work with businesses and deal with quantity than it is 
 to deal with the public or end customer and all the hassles with that.  I'm 
 positive that AudioCodes doesn't want to staff a customer service department 
 to deal with Joe Sixpack who's cousin's friend's son is a computer whiz and 
 hooked up this phone over teh intarweeb thingie for him but he just can't 
 it working perfect.

I never said they did, I replied to the notion that $10/license isnt
that much and people should just pay digium because we owe them for beta
testing the software for them that they sell commercially.  I used a
specific example designed specifically to show that the $10/lciense fee
could actually be a considerable sum instead of only $10 which is what I
replied to.  I am now begining to think that you didnt follow the thread
or even read what I replied to.  Out of curiosity do you read slashdot?


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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Re: Fwd: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Jon Lewis


On 6/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:


- Sahil Gupta [EMAIL PROTECTED] wrote:

We recently had around 60-80 licenses become useless because Digium
refused to renew the keys on that.  That was a bit of money kissed
goodbye.


Unless you had been clearly abusing the key licensing system, our
support department will never refuse to enable a new registration on
your license key(s). There is no 'renew the keys', though, since they
don't expire.


I hope that's the actual official policy now.  There seems to have been 
some internal conflict or communications failure at Digium a few months 
ago as to whether or how many times a g729 license key can be reset.


As a service provider (you could call us an Asterisk ASP), we regularly 
build  host systems for customers, retire/upgrade systems, swap out 
hardware, add interfaces, etc. which causes problems with the g729 
licensing.


In one attempt a few months ago to get a license reset, I was initially 
told it was now policy that Digium would only reset the registration count 
once, and after that, you were SOL (or forced to play MAC address changing 
games or as someone else posted, try hacking around the license key code).


In that particular case, the customer's server had suffered a 2 disk RAID 
failure, and to get them back online, I moved them to a lower end system 
(what was readily available) while we waited for parts to get their dual 
xeon server back online.  Both motherboards had built-in dual ethernets.


IMO, locking the licensing to a piece of system thats often built-in, has 
been very annoying.  I think I'd be happier if it was locked to some sort 
of dongle (parallel, or more likely today, USB).  At least that way, we 
could easily move the key anytime we needed to.  It would be a bit of a 
pain any time a system needed to quickly be transfered to hardware already 
at another location.


The TRX idea sounds appealing, but I wonder how they'll handle servers 
that don't have internet access.  Not all VOIP servers are on the 
internet.


I've actually wondered if we could legally use Intel's code in cases where 
we have licenses bought from Digium, but they're not re-registerable 
because Digium wouldn't reset the use count.


--
 Jon Lewis   |  I route
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 Atlantic Net|
_ http://www.lewis.org/~jlewis/pgp for PGP public key_
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Re: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Woodoo People .pGa!

 Talk to digium about this on [EMAIL PROTECTED], they might be able to 
 help you out there.
 
 Zoa
 
 Chris Mason (Lists) wrote:
 
 I have no problem with paying Digium the $10 for G729 licenses, 
 everyone has to make money. It's the administration of the licenses 
 that sucks. I experiment with different hardware a lot, and make up 
 demo machines to install for customers with available hardware. I have 
 to put G729 licenses on them, usually $100 each time, and when I 
 install the real hardware for the client, I can't transfer the 
 licenses. If I scrap that machine or change the interfaces, that's a 
 $100 loss. I believe when you buy a number of licenses, that should 
 determine how many instances you can use, regardless of how you want 
 to deploy them.
 In short, the method of enforcement is poor and leads to resentment 
 from customers. Surely Digium can construct a better system?

i think, for those of us, who would like to transfer licences from one box
to other (i mean more than 1-2 or 10), we would have to buy a hardware
base lock (of course, i don't care about, if the lock would contact
digium once a day or so) like usb, or a dumb pci ethernet card, so
if we need we can move it to other. what do you think?
(sadly there is no a 7day demo licence or anything to test) 
-- 
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] Prices of g729 codec

On Mon, 2006-06-05 at 10:00 -0400, Andrew Kohlsmith wrote:
 On Saturday 03 June 2006 04:05, Sahil Gupta wrote:
  We recently had around 60-80 licenses become useless because Digium
  refused to renew the keys on that.  That was a bit of money kissed
  goodbye.
 
 Ok, that's a great fairy tale.  Now tell us the true story.
 

Well it may be it maybe not, I wouldnt call people liars without proof
however.  What I will do is state that I have written a tool that allows
you to port digium licenses to different boxes.  Lets say that digium is
closed (2am, weekend, whatever) and your box caught on fire or whatever.
You cant get them ported right then because digium is closed.  My tool
will let you convert that license file to something else.

I will be making this program available publicly this week pending a
final test to ensure that it doesnt itself cause your system to catch on
fire :)


 When you buy licenses from Digium, you register them and they are branded 
 with information from your machine (most likely MAC address of the NIC, but 
 I'm not 100% sure nor do I particularly care for the details.)
 

And I am relying on that fact to thwart piracy with my tool.  If the
license file is shared and later discovered to be shared it will be easy
to track who leaked it and thus cause fewer if any people to leak it.
That and the fact that if people pay for the digium codec they are doing
it for the license not the codec itself since there are unlicensed ones
out there if they dont want the license.  The only reason to get the
digium codec is infact the license that accompanies it, so piracy on any
level should be rare if at all.


 If you upgrade hardware you may have to re-register them.  Digium allows you 
 to automatically re-register once without phone calls or any explanation.  
 After that, you cannot re-register without calling Digium and making a case 
 for it.  This is a restriction placed on them by the patent holders of the 
 g.729 codec.
 
not really, that is conjecture.  Having entered into a g729 license
myself I can attest that changing the license like that isnt a
requirement of my contract.  


 So the TRUE story is that you had 60-80 licenses, registered, changed your 
 hardware, re-registered, changed your hardware AGAIN and for one reason or 
 another failed to convince Digium that it was a legitimate change to warrant 
 a re-registration.
 
That may be, which goes with what he said that you said was a faery
tale.  So which is it, faery tale or fact?  You seem to be inconsistant.
And if he could have made a case that he needed it changed, wouldnt that
negate your argument that the patent holders wont let digium change it
more than once?  Is iut the patent holders (or their authorized agent
sipro.com as the case may be) or digium that has the discretion?


 I have personally called Digium and provided sufficient reasoning to grant me 
 a third registration.  
 
 So honestly now, Sahil, what did you guys do that was so different?  It 
 *really* pisses me off when people like you give a half-assed, half-baked 
 digium sucks post.  If you've got an honest beef with Digium, then sure, 
 lay it all out, but don't present half the fucking story and then bitch about 
 how the big bad Digium beat you up and stole your lunch money.
 
 ... just like my kids...  Wh, Joshua hit me!  Yeah but you've been 
 bugging the shit out of him for the last 5 minutes and he asked you nicely to 
 stop twice.  I'd have hit you too, Katie.

I cant speak for you hitting your kids when they bother you for 5
minutes, however if it bothers you that people post 'I had a problem
with digium' email and your explanation contradicts itself, even your
claim that someone is being less than honest when you yourself state
later that they were being honest seems dubious at best.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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Re: [Asterisk-Users] PHP-AGI help

2006-06-05 Thread Lewis Agosta

Yes. Jon is correct.

$agi[str_replace(agi_,,$s[[0])] = trim($s[[1]);

This line needs some work... Your brackets are mismatched.
On 6/2/06, Jon Farmer [EMAIL PROTECTED] wrote:
Yes you have a parse error in your PHP when I saved it locally and run it from the command line I got
syntax error, unexpected '[', expecting ']' in test.php on line 33Jon FarmerTelford, Shropshire, UK- Original Message From: Matthew Warren [EMAIL PROTECTED]
To: asterisk-users@lists.digium.comSent: Friday, 2 June, 2006 3:32:10 PMSubject: [Asterisk-Users] PHP-AGI helpCan someone help me with this AGI script to send an email.It just isn't
working.The file is being called in the dialplan and is saved as em.agibut it isn't sending the email.#!/usr/bin/php4 -q?phpob_implicit_flush(true);set_time_limit(6);$in = fopen(php://stdin,r);
$stdlog = fopen(/var/log/asterisk/my_agi.log, w);// toggle debugging output (more verbose)$debug = false;// Do function definitions before we start the main loopfunction read() {
global $in, $debug, $stdlog;$input = str_replace(\n, , fgets($in, 4096));if ($debug) fputs($stdlog, read: $input\n);return $input;}function errlog($line) {
global $err;echo VERBOSE \$line\\n;}function write($line) {global $debug, $stdlog;if ($debug) fputs($stdlog, write: $line\n);echo $line.\n;
}// parse agi headers into arraywhile ($env=read()) {$s = split(: ,$env);$agi[str_replace(agi_,,$s[[0])] = trim($s[[1]);if (($env == ) || ($env == \n)) {
break;}} $sender = [EMAIL PROTECTED]; $recipient = [EMAIL PROTECTED];
 $subject = call from someone; $header = From:  . $sender . \r\n; $header.= Reply-to:  . $sender . \r\n; mail($recipient, $subject, $message, $header);
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Re: [Asterisk-Users] Prices of g729 codec

On Mon, 2006-06-05 at 10:46 -0400, Paul wrote:
 I really doubt that Digium would insist on the $10 fee for a quantity buyer.

no they do give some discount for quantity, people have mentioned that
when they bought a bunch.  However I think  they said it was close to
$8/license for 672 channels.  Not a whole lot of a discount and
certainly more than other solutions.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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[Asterisk-Users] SpanDSP and analog Digium channels (TDM400P)

2006-06-05 Thread Raul Fragoso


Hi

I am trying to use Asterisk as a backend to send and receive faxes over 
analog channels connected to a Siemens HiPath 3550 switch and a TDM400P 
card.
Receiving faxes, including multipage ones, works really fine, we have no 
issues at all. But when it comes to send faxes using the app_txfax 
application, spandsp can't send multipage TIFF files for some reason, it 
stops sending in the middle or end of the first page. To isolate the 
problem, I have tested sending faxes with two different fax machines 
over the same analog channels, and they work 100%, so I don't think it 
is an issue with the switch settings. I have also measured the optimal 
settings for TX and RX gain for the zaptel side, and tried many 
different levels without success. I have disabled echo cancelling too.
Now I am stuck in a dilemma, whether it is an issue with spandsp itself 
or zaptel affecting the audio transmission and consequently the 
transmitted FAX frames.
My question is, have anyone tested sending multipage faxes using spandsp 
with analog zaptel channels ? If so, is there anything that I should be 
aware of to fix this issue ?
I'm currently using libtiff 3.8.2 (but tested without success with 3.7.1 
and 3.8.0 too), spandsp 0.0.2pre26 (tested with pre23 and pre25 with 
same results), zaptel 1.2.5 and asterisk 1.2.7.1.


Thanks in advance for your time.

Regards,

Raul
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Re: [Asterisk-Users] Prices of g729 codec

On Monday 05 June 2006 10:46, Paul wrote:
 I really doubt that Digium would insist on the $10 fee for a quantity
 buyer.

I have no idea (I do not work for Digium) but if you want to buy quantity 
g.729 codecs I'd be strongly looking at hardcore NAS equipment to do it for 
you.  The PC only has so much PCI bandwidth and PCIx/e or Infiniband TDM 
equipment isn't here.

-A.
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Re: Fwd: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Sahil Gupta


Hi,
I couldn't quite understand what was so wrong if someone was moving a bit 
of hardware around and requested key changes.  After all, the keys have 
been paid for and the registered person was requesting for the keys to be 
reset.


It was a while back... All good otherwise.

Regards,


Sahil Gupta
VoiceValley

On Mon, 5 Jun 2006, Jon Lewis wrote:


On 6/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:


- Sahil Gupta [EMAIL PROTECTED] wrote:

We recently had around 60-80 licenses become useless because Digium
refused to renew the keys on that.  That was a bit of money kissed
goodbye.


Unless you had been clearly abusing the key licensing system, our
support department will never refuse to enable a new registration on
your license key(s). There is no 'renew the keys', though, since they
don't expire.


I hope that's the actual official policy now.  There seems to have been some 
internal conflict or communications failure at Digium a few months ago as to 
whether or how many times a g729 license key can be reset.


As a service provider (you could call us an Asterisk ASP), we regularly build 
 host systems for customers, retire/upgrade systems, swap out hardware, add 
interfaces, etc. which causes problems with the g729 licensing.


In one attempt a few months ago to get a license reset, I was initially told 
it was now policy that Digium would only reset the registration count once, 
and after that, you were SOL (or forced to play MAC address changing games or 
as someone else posted, try hacking around the license key code).


In that particular case, the customer's server had suffered a 2 disk RAID 
failure, and to get them back online, I moved them to a lower end system 
(what was readily available) while we waited for parts to get their dual xeon 
server back online.  Both motherboards had built-in dual ethernets.


IMO, locking the licensing to a piece of system thats often built-in, has 
been very annoying.  I think I'd be happier if it was locked to some sort of 
dongle (parallel, or more likely today, USB).  At least that way, we could 
easily move the key anytime we needed to.  It would be a bit of a pain any 
time a system needed to quickly be transfered to hardware already at another 
location.


The TRX idea sounds appealing, but I wonder how they'll handle servers that 
don't have internet access.  Not all VOIP servers are on the internet.


I've actually wondered if we could legally use Intel's code in cases where we 
have licenses bought from Digium, but they're not re-registerable because 
Digium wouldn't reset the use count.


--
Jon Lewis   |  I route
Senior Network Engineer |  therefore you are
Atlantic Net|
_ http://www.lewis.org/~jlewis/pgp for PGP public key_
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[Asterisk-Users] More Level QueueSystem

2006-06-05 Thread Patrick Bök

Hi,

I am trying to set up a dial plan und I have a few problems to realise some
functions.

The dial plan should look like this:

123,1,Answer()
123,2,Queue(1stlevel,t)
123,3,Queue(2ndlevel,t)
123,4,Queue(3rdlevel,t)
123,5,Hangup()

If a member of the 1stlevel-Queue can answer the call it should be hanged up
after finishing. If not, the current member answering the call should be
able to transfer the caller to the 2ndlevel-Queue. And so on. How can I
check whether it is transfered or hanged up?

I do not know how to realise this workflow, the transfer, within the dial
plan and I have not found any solution within the Wiki.

The next problem I have got with the queue app is the value of the return
code:
0 for not being answered
-1 for hangup
1 for bridged (does bridge in this context mean the same as transfer???)

Would be nice if you could help me about the transfer problem between the
queues.

Thanks a lot,

Patrick


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Re: [Asterisk-Users] Prices of g729 codec

On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote:
  Again, 10k channels you'll have a half dozen MaxTNT boxes terminating
  DS3s. Your fixed costs will already be significantly higher and that
  little $10 license fee is included in that.

 Its not $10, which also goes along with something else I mentioned
 elsewhere.  Digium charges $10 but the max cost for a g729 license is
 about $1.25.  It goes down to about $0.10/license in quantity.  As such
 it doesnt add a whole lot to the cost of the device once the initial
 code is in place (as that development does have cost since the license
 fee doesnt cover any implementation, only the right to sell that
 implementation).

Stop the presses: quantity purchases get price breaks!  High enough quantities 
let you deal with the manufacturer directly!

This is news how?

I can buy a PIC16F877 for $13.23 in onesie-twosie quantities.  If I'm willing 
to buy them in 100 quantities I can get 'em for $7.32 apiece.  That's damn 
near 50% less.  If I commit to buying an entire reel (1200) of them, my price 
is $5.61.

Now let's stop fucking around and go directly to Microchip.  I want a mask-ROM 
PIC and commit to a minimum order of 1M pieces.  What do you think my price 
is?  I'm waiting for the email from their quoting department (and likely will 
get a we don't offer a masked ROM version of the 16F877, but I also asked 
for a masked ROM version of the PIC16C77, which I know they do make) but I'm 
willing to bet it'll be around $2 apiece.

What's my point?  If you're willing to deal in real volumes, the $10/transcode 
license fee doesn't apply.  You can either go directly to AudioCodes and 
negotiate a better fee ($1.25 is the number you're stating) or you have 
already paid the fee in fixed costs of the hardware you've got in order to be 
able to terminate that kind of call volume.

(and yes, you're right, the g.729 license cost on a MaxTNT isn't $10/port, 
that was a brainfart on my part.)

 And again to clarify, since this was aparently lost somewhere, I was
 responding to the mentality that everyone is a home user and its only
 $10 for a license and that is all anyone ever needs to pay.  You have
 proven me right, thanks again for that.  When you get out of the home
 user mindset the cost goes up dramatically and the argument that I
 responded to that the cost isnt that high at $10/license was invalid,
 even though you seem to be saying that it is that same cost, which
 anyone who really knows anything about the licensing knows that isnt
 true.

... So we're arguing the same point?

 testing the software for them that they sell commercially.  I used a
 specific example designed specifically to show that the $10/lciense fee
 could actually be a considerable sum instead of only $10 which is what I
 replied to.  I am now begining to think that you didnt follow the thread
 or even read what I replied to.  Out of curiosity do you read slashdot?

Your example was invalid, because no sane person running a business with that 
many concurrent calls will be transcoding them on PCs; they'll be terminating 
those calls directly to highly available, application-specific hardware whose 
per-port cost is significantly lower than anything you can hit on PC hardware 
at this time, while meeting better reliability and higher density levels than 
you can achieve on a PC platform at this point in time.  They'll have a 
number of g729 licenses on PCs for some small fraction of their port count 
for doing transcoding for specific purposes (call recording, etc.), but even 
at $10/license it's going to be a small cost of their overall fixed business 
costs.

I read Slashdot occasionally, yes.  Perhaps I've become infected.

-A.
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Re: [Asterisk-Users] Can´t send emails

David:thanks for your answer. I´m afraid is not the case.I can tell you this because I have an email server inside my network and it works fine.I can send emails from the [EMAIL PROTECTED] to itself (the root) but if I try to send an email to internet it gives me a time out.Thanks again.YrvingDavid K Parker [EMAIL PROTECTED] escribió: Many ISPs block outbound SMTP except directly to their mail servers. If this is the case, you could try using a mailhop service such as one provided by dnydns.org. On 6/5/06, yrving rivas [EMAIL PROTECTED] wrote: Hello everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly, so I can receive them directly from my phone with *97, but still the Asterisk don´t send it through Internet. As I have seen the SMTP is working, and I can ping any site of Internet, and any of my internal addresses, but still I have no idea about what is happening.Thanks.Yrving  __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - 
 http://correo.yahoo.com.mx/  ___--Bandwidth and Colocation provided by Easynews.com  --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users __Correo Yahoo!Espacio para todos tus mensajes, antivirus y
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Re: Fw: [Asterisk-Users] Compiling chan_bluetooth

2006-06-05 Thread Danko Miocevic

I use it all the time! now I can redirect calls to cell phones with no 
problems! I´ve tried a
SE T637 and a Motorola V3 and they worked really well. I´m still trying to 
receive calls from the
asterisk server to get into it from my cell phone, but I still can´t.. so 
I´m just able to

call other cell phones from internet.
Talking about prices, I just can tell you that here (Argentina) a cell phone 
like the
T637 costs about 100 U$S (new).. I think that you can get a bt cell phone 
for less than
that and you can get a BT USB adaptor for less than 30 U$S so.. maybe it´s 
cheaper (for me it is).



- Original Message - 
From: Woodoo People .pGa! [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Saturday, June 03, 2006 6:19 PM
Subject: Re: Fw: [Asterisk-Users] Compiling chan_bluetooth


does chan_bluetooth working well now? (integrating sound and signal 
channels

in BT?) If yes, it's better than using a cheap GSM adapter (like 150Euro)?

ps: i have tested it in last year with nokia6310, but with no luck.


Just to close the thread. The problem was that I was using an old version
of the code.
If anyone has the same problem, you can download the code from here:

http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz

Good luck,
   Danko




- Original Message - 
From: Danko Miocevic [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, May 30, 2006 8:48 PM
Subject: Re: [Asterisk-Users] Compiling chan_bluetooth


I´ve found a solution to my problem, I forgot to install the posix
development libraries.. now the error has dissapeared to
make place to a new error! :D I still can´t compile.
The new error says:

cc: chan_gsm_bt.o: no se usó el fichero de entrada del enlazador porque
no se hizo enlace
cc: -lbluetooth: no se usó el fichero de entrada del enlazador porque no
se hizo enlace

It says that the file wasn´t used because the linker didn´t make link...
don´t know what to do..
Any ideas?

Danko


- Original Message - 
From: Danko Miocevic [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, May 27, 2006 12:51 PM
Subject: [Asterisk-Users] Compiling chan_bluetooth


Hello, I´m trying to use my phone with asterisk to get GSM connectivity
but I can´t compile the code.
I got the asterisk, chan_bluetooth, zaptel and libpri sources, the last
two ones compiled perfectly.
I have added this to the /usr/src/asterisk/channels/Makefile:
   include /usr/src/chan_bluetooth/Makefile
and I´ve also added chan_bluetooth.so to the CHANNEL_LIBS var.
When I do make install in the asterisk directory I get lots of this
error:

/usr/src/chan_bluetooth/chan_bluetooth.c:2469: error: dereferencing
pointer to incomplete type

and some others like:

/usr/src/chan_bluetooth/chan_bluetooth.c: En la función
`remove_sdp_records':
/usr/src/chan_bluetooth/chan_bluetooth.c:2937: error: `sdp_session_t'
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2937: error: `sdp' undeclared
(first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2938: error: `sdp_list_t'
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2938: error: `attr' undeclared
(first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2939: error: `sdp_record_t'
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2939: error: `rec' undeclared
(first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2948: aviso: implicit
declaration of function `sdp_connect'
/usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `BDADDR_ANY'
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `BDADDR_LOCAL'
undeclared (first use in this function)
/usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: 
`SDP_RETRY_IF_BUSY'

undeclared (first use in this function)

I´m working on a stable debian, kernel 2.6 and my asterisk is 1.2. I
really don´t know what is happening, if someone
has an idea I´d be glad to hear it. Thanks for reading,

Danko
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Re: [Asterisk-Users] Can´t send emails

Alex:I verified my SMTP with "telnet ([EMAIL PROTECTED] ip) 25" and sent an email to root.I have another mail server inside my network and, as happens trying to send an email to internet, it gives me a time out.I can give you all the details if you would.Thanks for your help.yrvingAlex Robar [EMAIL PROTECTED] escribió: Yrving,How have you verified that your SMTP is working? Can you actually send a message from your Asterisk server to one of your internal addresses?AlexOn 6/5/06,  yrving rivas [EMAIL PROTECTED] wrote: Hello
 everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly, so I can receive them directly from my phone with *97, but still the Asterisk don´t send it through Internet. As I have seen the SMTP is working, and I can ping any site of Internet, and any of my internal addresses, but still I have no idea about what is happening.Thanks.Yrving  __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya -  http://correo.yahoo.com.mx/ 
 ___--Bandwidth and Colocation provided by Easynews.com  --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  
 http://lists.digium.com/mailman/listinfo/asterisk-users __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/ ___
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Re: [Asterisk-Users] Prices of g729 codec

On Monday 05 June 2006 11:12, trixter aka Bret McDanel wrote:
 Well it may be it maybe not, I wouldnt call people liars without proof
 however.  What I will do is state that I have written a tool that allows

Fair enough.

 That may be, which goes with what he said that you said was a faery
 tale.  So which is it, faery tale or fact?  You seem to be inconsistant.

How am I being inconsistent?

 And if he could have made a case that he needed it changed, wouldnt that
 negate your argument that the patent holders wont let digium change it
 more than once?  Is iut the patent holders (or their authorized agent
 sipro.com as the case may be) or digium that has the discretion?

I didn't say that the patent holder won't let Digium change it more than once.  
I believe I said something along the lines of without good reason which is 
a statement I stand by.  As far who has the discretion... I do not know.  I 
assume (yes, a dangerous passtime) that somewhere within the Digium/sipro 
agreement there is the understanding that Digium will make some effort to 
prevent abuse of the system.

 I cant speak for you hitting your kids when they bother you for 5
 minutes, however if it bothers you that people post 'I had a problem
 with digium' email and your explanation contradicts itself, even your
 claim that someone is being less than honest when you yourself state
 later that they were being honest seems dubious at best.

Again, where is my inconsistency?  You seem like a bright enough lad to 
identify a sarcastic and exaggerated example (so much so that you turn it 
into a thinly veiled personal attack on my parenting skills), so please show 
me where I'm inconsistent.

-A.
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[Asterisk-Users] Configuring behaviour of flash hook

2006-06-05 Thread Henry Margies

Hi all,

I have some problems configuring the right behaviour of Zaptel devices
in asterisk. Especially with three way call, call waiting and call
transfers.

Having one party on hold and the other on line I would like to have
 - Flash Hook + 3 to activate three way call
 - Flash Hook + 2 to swap between user on hold and user on line.

On call waiting I would like to have:
 - Flash Hook + 2 to accept waiting call 
 - Flash Hook + 0 to reject it

Having one party on hold while talking with an other I would also like
to have:
 - Flash Hook + 1 to disconnect from the current call.

I know that some of these features are configured in features.conf, but
always totally without the use of flash hook. 

Is there any way to configure this behaviour in asterisk?


Thanks in advance :)

Henry

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Re: [Asterisk-Users] RE: Size limitations of extensions.conf

2006-06-05 Thread Moises Silva


Asterisk support the concept of configuration engine, this means
that you can write a configuration engine to get the data from
anywhere. The default configuration engine is text_file_engine, that
reads the configuration from text files. This engine does not have any
limit in the code, so the only limit is the performance hit of
starting or reloading. Actually some limits exists for the size of
context names, nested includes etc, but no for number of lines.

Why dont use database engine? instead of large files?

Regards




On 6/5/06, Brent Torrenga [EMAIL PROTECTED] wrote:

If you need to do a couple differing operations on a list of many
area/country codes, then you may consider using the database to let the dial
plan choose what to do, rather than go through so many extensions.

I mention this to keep your extensions.conf easier to read, not because I
know whether or not a long extensions.conf will break things...

 Can someone tell me the size (or any other) limitations for the
extensions.conf?

 We have managed to keep our file pretty small thanks to AGI but we are
 about to setup a bunch of call restrictions based on area and country
 code.

 One line per area code in the US alone adds a LOT of text to this file.

 Is it a bad thing to have 5 or 6000 lines of text in your
 extensions.conf on a production system?

 Will it affect the performance?


Sincerely,

Brent A. Torrenga

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:[EMAIL PROTECTED]
web:www.torrenga.com

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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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[Asterisk-Users] Mixing meetme conferences

2006-06-05 Thread Erick Perez


Have anyone experienced mixed meetme conferences?
Im running a 12 seat call center outbound only. Asterisk 1.2.8,
SIP/ulaw at the phones, SIP/ulaw to the SIP terminator.

Machine: pentium Dual core 2.0ghz (533 fsb) , 1 GB RAM (533mhz ddr) ,
2 SATA 80GB DISK in RAID 0 via software RAID driver. Two Intel PRO
10/100 NIC, IRQs are separated, an X100P so not to use ztdummy.
Motherboard is an Intel 945GNT.


output of vmstat
procs ---memory-- ---swap-- -io --system-- cpu
r  b   swpd   free   buff  cache   si   sobibo   incs us sy id wa
2  0  0 493284  44996 29092400 216  247   116  3  2 95  0


cat /proc/interrupts

  CPU0   CPU1
 0:   24778955   24730655IO-APIC-edge  timer
 1:  8  0IO-APIC-edge  i8042
 8: 76 90IO-APIC-edge  rtc
 9:  0  0   IO-APIC-level  acpi
12: 66  0IO-APIC-edge  i8042
14:  88370  86773IO-APIC-edge  ide0
15:  74129  72701IO-APIC-edge  ide1
169:  0  0   IO-APIC-level  uhci_hcd
185: 117692208   IO-APIC-level  eth1, uhci_hcd
193:  0  0   IO-APIC-level  uhci_hcd
201:  0  0   IO-APIC-level  uhci_hcd
209:   24772638   24706144   IO-APIC-level  wcfxo
217:4126715  0   IO-APIC-level  eth0
NMI:   49705668   49705619
LOC:   49515422   49526061
ERR:  0
MIS:  0


--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones para centros de datos
Panama, Republica de Panama

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Re: [Asterisk-Users] Prices of g729 codec

On Mon, 2006-06-05 at 12:01 -0400, Andrew Kohlsmith wrote:
 On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote:
   Again, 10k channels you'll have a half dozen MaxTNT boxes terminating
   DS3s. Your fixed costs will already be significantly higher and that
   little $10 license fee is included in that.
 
  Its not $10, which also goes along with something else I mentioned
  elsewhere.  Digium charges $10 but the max cost for a g729 license is
  about $1.25.  It goes down to about $0.10/license in quantity.  As such
  it doesnt add a whole lot to the cost of the device once the initial
  code is in place (as that development does have cost since the license
  fee doesnt cover any implementation, only the right to sell that
  implementation).
 
 Stop the presses: quantity purchases get price breaks!  High enough 
 quantities 
 let you deal with the manufacturer directly!
 
 This is news how?
 
again you are either intentionally ignoring what is said or unable to
read, I dont know or care.  You cant despite your claims go directly to
sipro.com (the only authorized agent from the g729 consortium for
licensing) if you are an end user.  No matter what quantity you want to
get.  So your whole tirade misses the point.  Again.


 What's my point?  If you're willing to deal in real volumes, the 
 $10/transcode 
 license fee doesn't apply.  You can either go directly to AudioCodes and 
 negotiate a better fee ($1.25 is the number you're stating) or you have 
 already paid the fee in fixed costs of the hardware you've got in order to be 
 able to terminate that kind of call volume.
 

audiocodes doesnt sell the licenses, Sipro lan telecom inc does.  I
wonder if that is where you went for your proof earlier that digium cant
(despite kevins statement that digium does) change the licenses more
than once.


 ... So we're arguing the same point?
 
I dont know what you are trying to say, you keep commenting on stuff I
am not saying.  


 Your example was invalid, because no sane person running a business with that 
 many concurrent calls will be transcoding them on PCs; they'll be terminating 

its invalid becuase everyone that runs a business of any size is sane?
I disagree, but will let that go.  Of course if the opening statement is
invalid what does that say about the counter argument?

You further are STILL ignoring what the context was that the original
post was in reply to, and the reply to attempt to correct your bad
information and in some cases even flat out wrong information (as
disputed by digium employees).  But hey you are entitled to your own
delusions.  You seem bent on proving me wrong even if that means
misquoting, lying, making stuff up, or just being delusional.  So I will
let you have that victory, after all if you are willing to fight this
hard to be right you must not be able to be right that often.

You win I am wrong, digium is wrong and the g729 authorized agent (the
only one) is wrong.  

I will even concede to the fact that the context I originally intended
when I replied was wrong and that it was the undisclosed one that you
brought up later.

you win, let it go

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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Re: [Asterisk-Users] Can´t send emails

2006-06-05 Thread Lewis Agosta

Have you checked the validity of the mailcmd setting in voicemail.conf? Make sure that the command supplied there is the exact location to the application that will process your mail.

ie. mailcmd=/usr/sbin/sendmail -t

Also, make sure that the variable emailbody doesn't have something funky defined that is causing it to error out. Have you reviewed your mail server logs to see if the email is being sent? You would most likely get some good information from there if you could determine if the email is being sent out, or erroring out during the attempt.

On 6/5/06, yrving rivas [EMAIL PROTECTED] wrote:

Alex:I verified my SMTP with telnet ([EMAIL PROTECTED] ip) 25 and sent an email to root.I have another mail server inside my network and, as happens trying to send an email to internet, it gives me a time out.
I can give you all the details if you would.Thanks for your help.yrvingAlex Robar 
[EMAIL PROTECTED] escribió: 
Yrving,How have you verified that your SMTP is working? Can you actually send a message from your Asterisk server to one of your internal addresses?
Alex

On 6/5/06, yrving rivas [EMAIL PROTECTED]
 wrote: 

Hello everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly, so I can receive them directly from my phone with *97, but still the Asterisk don´t send it through Internet. 
As I have seen the SMTP is working, and I can ping any site of Internet, and any of my internal addresses, but still I have no idea about what is happening.Thanks.
Yrving

__Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ˇgratis! Regístrate ya - 
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Re: [Asterisk-Users] RE: Size limitations of extensions.conf

On Mon, 2006-06-05 at 11:24 -0500, Moises Silva wrote:
 Asterisk support the concept of configuration engine, this means
 that you can write a configuration engine to get the data from
 anywhere. The default configuration engine is text_file_engine, that
 reads the configuration from text files. This engine does not have any
 limit in the code, so the only limit is the performance hit of
 starting or reloading. Actually some limits exists for the size of
 context names, nested includes etc, but no for number of lines.
 
 Why dont use database engine? instead of large files?
 
 Regards

That doesnt solve the root problem.  The configuration engine would be
called at startup/reload and load everything into memory.  A better
approach might be what some have done in private patches to read in
realtime each customers and only each customers information on a per
needed basis.  

Lets say you have a 'cluster' of asterisk servers that all feed off the
same database and in total you have several thousand hosted customers.
Each customer has many lines, possibly several hundred lines that would
form their extensions.conf entries.  

If you load that into memory you are consuming a lot of memory on each
box of your 'cluster' that will mostly be unused because not all of your
customers will receive calls on all the same systems.  Sifting through
all of this the way that is done also has a certain cost (although I
dont know the actual cost of doing it in a DB).  

If you load each customers configuration on demand you lower the memory
footprint and add the ability to have easier updates all at the same
time.  You can do some of this with realtime dialplans, but you cant get
the faster processing available that the one patch that does all of this
gets.

The way the dialplan works has its own problems (internally to asterisk)
with large extensions.conf.  For reference the one I am talking about
would be close to 200M if printed out, and a real pain to manage via any
of the standard 'engines'.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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[Asterisk-Users] Asterisk chroot

I thought I saw a guide at voip-info that described how to set up and asterisk 
to run in a chrooted environment. Now, I can't seem to find it. Anyone know 
where such a guide may be?

Doug
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Re: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Steve Underwood


Andrew Kohlsmith wrote:


On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote:
 


Again, 10k channels you'll have a half dozen MaxTNT boxes terminating
DS3s. Your fixed costs will already be significantly higher and that
little $10 license fee is included in that.
 


Its not $10, which also goes along with something else I mentioned
elsewhere.  Digium charges $10 but the max cost for a g729 license is
about $1.25.  It goes down to about $0.10/license in quantity.  As such
it doesnt add a whole lot to the cost of the device once the initial
code is in place (as that development does have cost since the license
fee doesnt cover any implementation, only the right to sell that
implementation).
   



Stop the presses: quantity purchases get price breaks!  High enough quantities 
let you deal with the manufacturer directly!


This is news how?

I can buy a PIC16F877 for $13.23 in onesie-twosie quantities.  If I'm willing 
to buy them in 100 quantities I can get 'em for $7.32 apiece.  That's damn 
near 50% less.  If I commit to buying an entire reel (1200) of them, my price 
is $5.61.


Now let's stop fucking around and go directly to Microchip.  I want a mask-ROM 
PIC and commit to a minimum order of 1M pieces.  What do you think my price 
is?  I'm waiting for the email from their quoting department (and likely will 
get a we don't offer a masked ROM version of the 16F877, but I also asked 
for a masked ROM version of the PIC16C77, which I know they do make) but I'm 
willing to bet it'll be around $2 apiece.
 

If you'll pay $2 for a million off of the PIC16C77 you must be a very 
very rich main. Anything over $0.70 would be a serious ripoff.


Regards,
Steve

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Re: [Asterisk-Users] Prices of g729 codec

On Monday 05 June 2006 12:29, trixter aka Bret McDanel wrote:
 you win, let it go

I'm not looking to win anything here.  

I got into this thread because of Brian and Lee's dialogue on Friday.  What 
set me off was Lee's yes, be a good colonist and don't dump any more tea 
into the harbour.

You came in and said that $10 gets only one transcode license, and asked what 
should be done if you're Vonage-sized and need a million licenses.  You then 
toned it down a little and asked what to do if you were a smaller company and 
only needed 10k licenses.

My response to that was that for either one of those cases you would be insane 
to be transcoding that many conversations on PC hardware.  If you have 10k 
PSTN channels I posited that you should have sense enough to be using 
carrier-grade hardware to terminate to TDM instead of trying to make it on 
racks of PCs and using general processors to do the work where cheaper, 
faster and denser solutions could do it.

You then stated that If you are going to bring business into it, at least 
accept that a business would most likely pay more than $10 for their 
licensing needs.

Of *course* that is correct, the statement's so obvious that I responded by 
saying if you're going to be using large numbers of g729 licenses it behooves 
you to investigate other options, such as carrier-grade NAS hardware that not 
reduce your port cost but include the g729 license.

Your entire argument seems to be If I'm a business and I need g.729, it's 
very likely going to cost me more than $10.

Is that fair description of your argument?  Or am I misreading?

-A.
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[Asterisk-Users] Looking for postpaid quality A-Z termination

2006-06-05 Thread Jean-Michel Hiver


Hi List,

After quite a bit of struggle, it looks like I'm all ready to roll out 
prepaid cards on my small island. I now have a 4 E1s with a bit of spare 
capacity in order to accept incoming calls, and I can route Reunion 
Island mobile and fix through my own installations.


For all other destinations, I need a carrier. I need good wholesale 
prices to Comoros, Mauritius, Madagascar, India, China / HK, France, UK.


I am looking for some quality A-Z  SIP / g.729 termination providers who 
are willing to work on a postpaid basis. This is because I have enough 
work as it is, and I don't want to be checking for account balance all 
the time. Since I work on a postpaid basis with my own clients (for VoIP 
termination), it would fit my business model better.


I am going to start selling the cards very progressively, so don't 
expect volume to be very high straight away. Since I want to work with 
postpaid providers, this is a good thing since it will give us time to 
build some trust as volumes progressively go up.


Eventually I plan to be selling under 12 month around 1,000 cards per 
week, which would amount to roughly 1,500 EUR worth of call termination 
(weekly).


I understand that doing postpaid is considered risky by many. However I 
have been operating in the VoIP industry for a year and a half and have 
a profitable and cash flow positive business, with good cash reserves. I 
invite you to check the archives to see that I've been around doing VoIP 
for a little while now.


If you are interested in building this business relationship with me, 
please let me know.


Best Regards,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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RE: [Asterisk-Users] RE: Size limitations of extensions.conf

If by database you are referring to an external database, such as MySQL, you 
have to address failover, redundancy and performance issues if you go in that 
direction.

 -Original Message-
 From: Moises Silva [mailto:[EMAIL PROTECTED]
 Sent: Monday, June 05, 2006 10:24 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] RE: Size limitations of extensions.conf

 Asterisk support the concept of configuration engine, this means
 that you can write a configuration engine to get the data from
 anywhere. The default configuration engine is text_file_engine, that
 reads the configuration from text files. This engine does not have any
 limit in the code, so the only limit is the performance hit of
 starting or reloading. Actually some limits exists for the size of
 context names, nested includes etc, but no for number of lines.

 Why dont use database engine? instead of large files?

 Regards

 On 6/5/06, Brent Torrenga [EMAIL PROTECTED] wrote:
  If you need to do a couple differing operations on a list of many
  area/country codes, then you may consider using the 
 database to let the dial
  plan choose what to do, rather than go through so many extensions.

  I mention this to keep your extensions.conf easier to read, 
 not because I
  know whether or not a long extensions.conf will break things...

   Can someone tell me the size (or any other) limitations for the
  extensions.conf?

   We have managed to keep our file pretty small thanks to 
 AGI but we are
   about to setup a bunch of call restrictions based on area 
 and country
   code.

   One line per area code in the US alone adds a LOT of text 
 to this file.

   Is it a bad thing to have 5 or 6000 lines of text in your
   extensions.conf on a production system?

   Will it affect the performance?

  Sincerely,

  Brent A. Torrenga

  Torrenga Engineering, Inc.
  907 Ridge Road
  Munster, Indiana 46321-1771

  tel:+1 219 836 8918 x325
  fax:+1 219 836 1138
  email:[EMAIL PROTECTED]
  web:www.torrenga.com

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 -- 
 Su nombre es GNU/Linux, no solamente Linux, mas info en 
http://www.gnu.org;
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Re: [Asterisk-Users] reinvite


- Osama Kamal [EMAIL PROTECTED] wrote:
 does thia apply on SIP only or also IAX?

It has nothing to do with the VOIP protocol, it is the nature of how NAT and 
port translation works.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] fine-tuning asterisk questions


- William Piper [EMAIL PROTECTED] wrote:

 By gone forever in 1.6... do you mean that even the j in the dial
 plan won't work either? Will it just go to the next priority in the
 event of a congested or busy signal?

That is correct. All the 'j' options will go away, in favor of channel-variable 
result codes returned by the applications.

 I assume goto will still work... right?

Uhh... yeah. That would be silly to remove it :-)

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Configuring behaviour of flash hook

2006-06-05 Thread Eric \ManxPower\ Wieling

I don't think you can do what you want.  The Zap custom calling features 
work very much like Centrex service in the USA.


Henry Margies wrote:

Hi all,

I have some problems configuring the right behaviour of Zaptel devices
in asterisk. Especially with three way call, call waiting and call
transfers.

Having one party on hold and the other on line I would like to have
 - Flash Hook + 3 to activate three way call
 - Flash Hook + 2 to swap between user on hold and user on line.

On call waiting I would like to have:
 - Flash Hook + 2 to accept waiting call 
 - Flash Hook + 0 to reject it


Having one party on hold while talking with an other I would also like
to have:
 - Flash Hook + 1 to disconnect from the current call.

I know that some of these features are configured in features.conf, but
always totally without the use of flash hook. 


Is there any way to configure this behaviour in asterisk?


Thanks in advance :)

Henry

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--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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RE: [Asterisk-Users] Config Revision Control

 -Original Message-
 From: Michiel van Baak [mailto:[EMAIL PROTECTED]
 Sent: Monday, June 05, 2006 8:03 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Config Revision Control

 On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote:
  On Saturday 03 June 2006 02:47, Michiel van Baak wrote:
   I use subversion for this. Every server has its own branch.
   There's also a branch called 'common'
   All the server specific branches are svn-copied and svnmerge
   init from this branche.
   Then the svn automerge thingie Kevin wrote for the asterisk
   svn tree is automerging changes to the 'common' tree to all
   the server trees.
   In the server trees I make changes specific for one server.

  Can you give some more details?  I am VERY interested in this!

 Most is already in my previous mail.

 This is my layout:
 branches/common
 branches/servers/home001
 branches/servers/home002
 branches/servers/cust001

 Like that, you get the idea
 The branches/common holds a full config, cept for sip users etc. So
 all the [global] and [default] stuff. Also the
 extensions.conf has some macro's and contexts I need on
 every machine.

 The home001 etc hold the conf I actually run on a server.
 All the specific sip and iax peers/users are defined in it.
 Also the specific stuff for extensions.conf for that server.

 If I for example want the congestion in my default outbound
 routing macro to play congestion for 5 seconds instead of 10
 I only alter extensions.conf in branches/common
 The automerge will take care of the promoting it to all the
 other branches.

Hmmm. What do you do with other files such as AGI scripts, sound files, or 
music on hold?
Do you maintain separate trees for each of these? If you do, to completely 
update a system, don't you have to check out etc, agi, sound and moh all 
independantly?

Ideally it would be good if you could put it _ALL_ under a single tree, and 
then put Asterisk in a chrooted envionment. Then you could check out and update 
the configuration all in one go.

While I was playing with svn, it was driving me nuts. It would ALWAYS re-create 
the current directory, even if I said to check out all files from inside that 
directory. Means if you went to /etc/asterisk and checked out asterisk, you'd 
get /etc/asterisk/asterisk. Yuk.

Doug.

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Re: [Asterisk-Users] Duplicate CDRs


- Mark Drayton [EMAIL PROTECTED] wrote:

 Okay. Which one writes the 18-field line (with uniqueid and
 userfield)?cdr_custom.conf has fields for these two but the wiki docs
 also say that cdr_csv will write uniqueid and userfield if configured.
 Can I just unload whichever one I don't need to stop it writing or do
 I need to reload the logging/cdr system? Presumably cdr_csv that
 writes accountcode.csv as cdr_custom specifies Master.csv.

It would have taken you far less time to just try it instead of composing a 
lengthy question... especially since cdr_custom.conf shows you _exactly_ the 
format it writes the records in, as it's a custom format.

The logging system is totally unrelated, as is cdr_manager (which is why I 
didn't mention it in my reply).

Simple answer: don't load cdr_csv, and configure cdr_custom.conf to write the 
records in the format you want.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Asterisk chroot

On Mon, 2006-06-05 at 10:44 -0600, Douglas Garstang wrote:
 I thought I saw a guide at voip-info that described how to set up and 
 asterisk to run in a chrooted environment. Now, I can't seem to find it. 
 Anyone know where such a guide may be?

http://www.voip-info.org/wiki-Asterisk+non-root

Regards,
Patrick

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Re: [Asterisk-Users] TDM-400 doesn't detect far-end hangup

2006-06-05 Thread Eric \ManxPower\ Wieling


Stephen Bosch wrote:

Hi:

I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with
kewlstart signalling.

When an outside caller calls the server, the TDM-400 goes off-hook and
provides a ringing tone to the caller. If the caller hangs up before the
receiving party answers the phone, Asterisk fails to detect the hang-up.
The TDM-400 stays off-hook, hogging the line, while Asterisk rings the
extension.

I thought the TDM-400 FXO module was supposed to detect far-end call drops.

Is there something I need to configure to make this work?


The telco has to drop battery.

--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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Re: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Martin Joseph


Girls, girls, you're both pretty...

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Re: [Asterisk-Users] Configuring behaviour of flash hook


- Henry Margies [EMAIL PROTECTED] wrote:

 I know that some of these features are configured in features.conf,
 but
 always totally without the use of flash hook. 

No, flash-hook features are handled by chan_zap itself.

 Is there any way to configure this behaviour in asterisk?

Not without changing the code in chan_zap, no. Making sequences of flash-hook 
plus a digit will be non-trivial, to say the least.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Prices of g729 codec


- Jon Lewis [EMAIL PROTECTED] wrote:

 IMO, locking the licensing to a piece of system thats often built-in,
 has 
 been very annoying.  I think I'd be happier if it was locked to some
 sort 
 of dongle (parallel, or more likely today, USB).  At least that way,
 we 
 could easily move the key anytime we needed to.  It would be a bit of
 a 
 pain any time a system needed to quickly be transfered to hardware
 already 
 at another location.

I have proposed that a number of times internally, only to be told (vehemently) 
that customers would never go for it. That includes responses from our 
distributors and channel partners, among others. It would also dramatically 
increase the cost for people buying one or two licenses, so it would have be an 
'alternate' registration means if it existed.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Looking for postpaid quality A-Z termination

2006-06-05 Thread Martin Joseph


What part of NON-COMMERCIAL do you not understand?

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Re: [Asterisk-Users] Mixing meetme conferences

2006-06-05 Thread Matt Florell


It would help if you included some more information, maybe like the
output from show channels concise from Asterisk and then a summary
of which channels and/or meetmes are mixing audio.

I have not run into any issues with audio from one meetme bleeding
into another, but since you mention meetme and call center I assume
you are using VICIDIAL which might mean you are having some issues
with your agent's not logging out properly.

MATT---

On 6/5/06, Erick Perez [EMAIL PROTECTED] wrote:

Have anyone experienced mixed meetme conferences?
Im running a 12 seat call center outbound only. Asterisk 1.2.8,
SIP/ulaw at the phones, SIP/ulaw to the SIP terminator.

Machine: pentium Dual core 2.0ghz (533 fsb) , 1 GB RAM (533mhz ddr) ,
2 SATA 80GB DISK in RAID 0 via software RAID driver. Two Intel PRO
10/100 NIC, IRQs are separated, an X100P so not to use ztdummy.
Motherboard is an Intel 945GNT.


output of vmstat
procs ---memory-- ---swap-- -io --system-- cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us sy id wa
 2  0  0 493284  44996 29092400 216  247   116  3  2 95  0


cat /proc/interrupts

   CPU0   CPU1
  0:   24778955   24730655IO-APIC-edge  timer
  1:  8  0IO-APIC-edge  i8042
  8: 76 90IO-APIC-edge  rtc
  9:  0  0   IO-APIC-level  acpi
 12: 66  0IO-APIC-edge  i8042
 14:  88370  86773IO-APIC-edge  ide0
 15:  74129  72701IO-APIC-edge  ide1
169:  0  0   IO-APIC-level  uhci_hcd
185: 117692208   IO-APIC-level  eth1, uhci_hcd
193:  0  0   IO-APIC-level  uhci_hcd
201:  0  0   IO-APIC-level  uhci_hcd
209:   24772638   24706144   IO-APIC-level  wcfxo
217:4126715  0   IO-APIC-level  eth0
NMI:   49705668   49705619
LOC:   49515422   49526061
ERR:  0
MIS:  0


--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones para centros de datos
Panama, Republica de Panama

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[Asterisk-Users] DTMF and DISA

2006-06-05 Thread Mr. Jones


Hi Folks,

I'm trying to test out Asterisk overall.

I'm having some problems with DTMF. Currently I'm playing with DISA,
but I'm worried this will happen when I get to implementing AAs etc.

I have a free SIP trunk from IPKall that I'm trying to make work.

I'm able to receive calls, and I've now setup and extension with DISA
and a password.

I connect ok from the PTSN, get the dialtone, and enter the password.

In the CLI I'm getting duplicate/extra/incorrect digits.

I've tried dtmfmode=auto, dtmfmode=inband, and dtmfmode=rfc2833 all
with similar results.

For testing I set the password to 67891 here's what I'm getting (small
sample size, but its pretty random):

app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078d00 got
bad password 677891
app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078cc8 got
bad password 6709
app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078cc8 got
bad password 677891

Any hints or tips are appreciated.

B
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RE: [Asterisk-Users] Config Revision Control

2006-06-05 Thread Aaron Daniel


While I was playing with svn, it was driving me nuts. It would ALWAYS re-create 
the current directory, even if I said to check out all files from inside that 
directory. Means if you went to /etc/asterisk and checked out asterisk, you'd 
get /etc/asterisk/asterisk. Yuk.

Doug.


Ahem.

cd /etc/asterisk
svn update


--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] Outgoing call bridging

2006-06-05 Thread mawali


Hi
Is there an easy way (without writing a C app) to make asterisk call 2 
numbers and then  bridge them into one conversatoin (preferably without 
using meetme).


Regards
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RE: [Asterisk-Users] Asterisk chroot

Thanks Patrick, but thats for non-root Asterisk, not chroot Asterisk.

Doug

 -Original Message-
 From: Patrick [mailto:[EMAIL PROTECTED]
 Sent: Monday, June 05, 2006 11:02 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Asterisk chroot

 On Mon, 2006-06-05 at 10:44 -0600, Douglas Garstang wrote:
  I thought I saw a guide at voip-info that described how to 
 set up and asterisk to run in a chrooted environment. Now, I 
 can't seem to find it. Anyone know where such a guide may be?

 http://www.voip-info.org/wiki-Asterisk+non-root

 Regards,
 Patrick

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Re: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Matt Florell


What are the reasons that people/companies/manufacturers use G729
instead of comperable codecs like GSM or Speex?

Microsoft and Apple both support GSM in their software, and Speex is
the same compression ratio as G729 yet is BSD-like licensed so no cost
whatsoever.

MATT---
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Re: [Asterisk-Users] Prices of g729 codec

On Monday 05 June 2006 13:03, Martin Joseph wrote:
 Girls, girls, you're both pretty...

But I'm prettier, right?  :-)

-A.
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[Asterisk-Users] Wanted: CISCO 186 ATAs

2006-06-05 Thread James Ching

Greetings,

I'm looking to purchase 10 Cisco 186 ATAs. Please send me quantities available, payment methodsand out the door pricing (shipping +tax +unit costs). 
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Re: [Asterisk-Users] Prices of g729 codec

On Monday 05 June 2006 13:05, Kevin P. Fleming wrote:
 I have proposed that a number of times internally, only to be told
 (vehemently) that customers would never go for it. That includes responses
 from our distributors and channel partners, among others. It would also
 dramatically increase the cost for people buying one or two licenses, so it
 would have be an 'alternate' registration means if it existed.

I too have proposed that to Digium, but without any kind of real response 
(this was quite a bit before you were hired, IIRC.)

I think that customers fall into two groups -- those who don't like additional 
hardware (too messy) and those who don't care so long as it just works.  
having both methods would certainly be beneficial, and could even be an 
additional revenue source... $10 for the licenses, $100 (or whatever) for the 
dongle, and you can transfer the licenses to the dongle.  Use libusb to 
communicate with it and it should be fairly portable as well.

JUST DO NOT MAKE A FUCKING PARALLEL PORT DONGLE.  USB that is either 
proprietary all the way, or USB which is USB-serial internally.  I'd also 
NOT recommend using a little USB PIC or Atmel part and writing your own 
license code...  there are companies whose business is this, and they could 
likely get your costs down significantly.

-A.
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[Asterisk-Users] This should be easy: What happens when the Calling Party hangs up

2006-06-05 Thread Julian Lyndon-Smith


svn trunk 31497

For the life of me, I can't get this :) I want to be able to catch the 
situation where the calling party hangs up *before* the call is 
connected to the called party. My dialplan is thus:


macro DialExternal(exten) {
Dial(Zap/G3/${exten},120,g,M(connected));
goto DialResult|r${HANGUPCAUSE}|1;
Hangup();
};

But the goto dialresult is not executed:

Executing [from-sip:1] Macro(SIP/7XX-b403, DialExternal|xx) in 
new stack
-- Executing [macro-DialExternal:1] Dial(SIP/7XX-b403, 
Zap/G3/07803034440|120|g|M([EMAIL PROTECTED])) in new stack

-- Requested transfer capability: 0x00 - SPEECH
-- Called G3/XX
-- Zap/10-1 is proceeding passing it to SIP/7XX-b403
-- Zap/10-1 is ringing
-- Hungup 'Zap/10-1'
  == Spawn extension (macro-DialExternal, s, 1) exited non-zero on 
'SIP/7XX-b403' in macro 'DialExternal'
  == Spawn extension (macro-DialExternal, s, 1) exited non-zero on 
'SIP/7XX-b403'


What do I need to put where ? I can catch all connected, calling party 
hangups and bad numbers etc


Julian

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[Asterisk-Users] porting g729 licenses to another machine

On Mon, 2006-06-05 at 12:05 -0500, Kevin P. Fleming wrote:
 I have proposed that a number of times internally, only to be told 
 (vehemently) that customers would never go for it. That includes responses 
 from our distributors and channel partners, among others. It would also 
 dramatically increase the cost for people buying one or two licenses, so it 
 would have be an 'alternate' registration means if it existed.
 

In addition there is the lag time between ordering and receiving the
dongle.  That could be worked around with a time expired software only
license that is good for say 10 days so you can get running right
away...  Dongles arent free and they would add a bit, likely $5-10, to
the total cost.  For 1-2 codecs not worth it.

However if you have a problem during digium off hours you can use the
generic example program I wrote (for other reasons it just appears to
work for this although I never tested it on a system with digium g729
codec installed nor did anyone else that I am aware of) will let you
change the MAC address only for the instance of asterisk that you
choose.  

This means that on a network view the system is using its real MAC addr,
for all programs except the ones you selected it would use the real MAC
addr, but for the one process you choose (my example uses ifconfig
becuase that is easy to instantly verify if it works) it will use an
alternate MAC addr.  

This is handy if you are on the other side of the world from digium and
digium is closed when your hardware fails.  The program itself was
written as an example of how to do this type of remapping, it just
appears to also work for this application.

It is my belief that this tool will not aid in piracy at all because
people that want a free g729 license will look at cisco (they run a
project which has an open g729 implementation for non-commercial
purposes) or the intel IPP stuff which is already ported to asterisk.
They are buying the digium modules to get the license since the codec is
already out there.  That means they want the licenses rather than a free
ride.  I also dont know for a fact that it works on the digium codecs,
only a hunch because there are only so many ways to get the MAC addr.

Now that you have read this far here is the link :)
http://www.0xdecafbad.com/Remapping-function-calls.html

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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Re: [Asterisk-Users] Prices of g729 codec

On Mon, 2006-06-05 at 13:47 -0400, Matt Florell wrote:
 What are the reasons that people/companies/manufacturers use G729
 instead of comperable codecs like GSM or Speex?
 
 Microsoft and Apple both support GSM in their software, and Speex is
 the same compression ratio as G729 yet is BSD-like licensed so no cost
 whatsoever.

speex isnt in all ATAs and other things.  So if its not there it offers
worse compression since the call wont go through :P


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com we pay you to terminate calls with us!


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Re: [Asterisk-Users] Can´t send emails

2006-06-05 Thread Alex Robar

Yrving,You can send to a local user on the system, but can you send to an external account? AlexOn 6/5/06, yrving rivas 
[EMAIL PROTECTED] wrote:Alex:I verified my SMTP with telnet (
[EMAIL PROTECTED] ip) 25 and sent an email to root.I have another mail server inside my network and, as happens trying to send an email to internet, it gives me a time out.I can give you all the details if you would.
Thanks for your help.yrvingAlex Robar [EMAIL PROTECTED]
 escribió: Yrving,How have you verified that your SMTP is working? Can you actually send a message from your Asterisk server to one of your internal addresses?
AlexOn 6/5/06,  yrving rivas 
[EMAIL PROTECTED] wrote: Hello
 everybody.I will apreciate your help in this case.I have the environment showed in the picture of the file included in this message.I would like to send trhough internet all the messages received by voicemail. I made all the configuration through the AMP, and all messages are saved properly, so I can receive them directly from my phone with *97, but still the Asterisk don´t send it through Internet. 
As I have seen the SMTP is working, and I can ping any site of Internet, and any of my internal addresses, but still I have no idea about what is happening.Thanks.Yrving 
 __Correo Yahoo!Espacio para todos tus mensajes, antivirus y antispam ˇgratis! Regístrate ya - 
 http://correo.yahoo.com.mx/ 
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http://correo.yahoo.com.mx/ 
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Re: [Asterisk-Users] DTMF and DISA

2006-06-05 Thread Eric \ManxPower\ Wieling

DTMF problems happen at the point where the PSTN call is converted to 
VoIP.  EXCEPT where you are using inband DTMF and ulaw or alaw codec. 
inband DTMF does not work with any other codec.


Mr. Jones wrote:

Hi Folks,

I'm trying to test out Asterisk overall.

I'm having some problems with DTMF. Currently I'm playing with DISA,
but I'm worried this will happen when I get to implementing AAs etc.

I have a free SIP trunk from IPKall that I'm trying to make work.

I'm able to receive calls, and I've now setup and extension with DISA
and a password.

I connect ok from the PTSN, get the dialtone, and enter the password.

In the CLI I'm getting duplicate/extra/incorrect digits.

I've tried dtmfmode=auto, dtmfmode=inband, and dtmfmode=rfc2833 all
with similar results.

For testing I set the password to 67891 here's what I'm getting (small
sample size, but its pretty random):

app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078d00 got
bad password 677891
app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078cc8 got
bad password 6709
app_disa.c:292 disa_exec: DISA on chan SIP/66.54.140.46-09078cc8 got
bad password 677891

Any hints or tips are appreciated.

B
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--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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RE: [Asterisk-Users] Config Revision Control

 -Original Message-
 From: Michiel van Baak [mailto:[EMAIL PROTECTED]
 Sent: Monday, June 05, 2006 8:03 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Config Revision Control

 On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote:
  On Saturday 03 June 2006 02:47, Michiel van Baak wrote:
   I use subversion for this. Every server has its own branch.
   There's also a branch called 'common'
   All the server specific branches are svn-copied and svnmerge
   init from this branche.
   Then the svn automerge thingie Kevin wrote for the asterisk
   svn tree is automerging changes to the 'common' tree to all
   the server trees.
   In the server trees I make changes specific for one server.

  Can you give some more details?  I am VERY interested in this!

 Most is already in my previous mail.

 This is my layout:
 branches/common
 branches/servers/home001
 branches/servers/home002
 branches/servers/cust001

 Like that, you get the idea
 The branches/common holds a full config, cept for sip users etc. So
 all the [global] and [default] stuff. Also the
 extensions.conf has some macro's and contexts I need on
 every machine.

 The home001 etc hold the conf I actually run on a server.
 All the specific sip and iax peers/users are defined in it.
 Also the specific stuff for extensions.conf for that server.

 If I for example want the congestion in my default outbound
 routing macro to play congestion for 5 seconds instead of 10
 I only alter extensions.conf in branches/common
 The automerge will take care of the promoting it to all the
 other branches.

 I use this script to do the automerging every hour:
 http://svn.digium.com/view/repotools/svn-automerge?rev=54view=markup
 This also means you have to use the modified svnmerge from
 the asterisk project:
 http://svn.digium.com/view/repotools/svnmerge?rev=63view=markup

 All my servers do auto svn up of the asterisk configs.

I guess this is wy beyond my knowledge of subversion. I just started 
playing with the directory structure I might use, and first thought was 
something like this:

[EMAIL PROTECTED] ~/cfg $ ls -l
total 16
drwxr-xr-x 2 dougg users 4096 Jun  5 12:24 acd
drwxr-xr-x 2 dougg users 4096 Jun  5 12:28 common
drwxr-xr-x 2 dougg users 4096 Jun  5 12:28 pbx
drwxr-xr-x 2 dougg users 4096 Jun  5 12:24 vm

where acd, pbx and vm refer to a function, or class of systems. pbx/ would have 
systems pbx1, pbx2 and pbx3 beneath it. Some files, such as sound files, and 
AGI are common to all systems, and hence the common/ directory. However, I have 
no idea what to do with it beyond that. I don't know how to push common changes 
out to all the other servers, or inherit, or whatever, or how to stop a common 
directory being created on the servers instead of putting the files from common 
under /var/lib/asterisk/agi-bin and /usr/lib/asterisk/sounds etc. Arrgh.

Doug.
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Re: [Asterisk-Users] Prices of g729 codec

2006-06-05 Thread Mike Fedyk


Kevin P. Fleming wrote:

- Jon Lewis [EMAIL PROTECTED] wrote:

  

IMO, locking the licensing to a piece of system thats often built-in,
has 
been very annoying.  I think I'd be happier if it was locked to some
sort 
of dongle (parallel, or more likely today, USB).  At least that way,
we 
could easily move the key anytime we needed to.  It would be a bit of
a 
pain any time a system needed to quickly be transfered to hardware
already 
at another location.



I have proposed that a number of times internally, only to be told (vehemently) 
that customers would never go for it. That includes responses from our 
distributors and channel partners, among others. It would also dramatically 
increase the cost for people buying one or two licenses, so it would have be an 
'alternate' registration means if it existed.
How hard is it to use a removable ethernet card for this type of usage?  
Also a USB ethernet if with Linux drivers should be usable for the 1U 
rackmount use case where all internal slots are in use.


Most of the complaints should be able to be remedied by and update in 
docs for recommended implementation (removable ethernet, either PCI or USB).


And just make sure the g729 codec can see those ethernet ports.
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Re: [Asterisk-Users] Wanted: CISCO 186 ATAs

2006-06-05 Thread Alex Robar

James,Please send this type of inquiry ONLY to the Asterisk-biz group, not the non-commercial discussion group.AlexOn 6/5/06, James Ching
 [EMAIL PROTECTED] wrote:
Greetings,

I'm looking to purchase 10 Cisco 186 ATAs. Please send me quantities available, payment methodsand out the door pricing (shipping +tax +unit costs). 

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[Asterisk-Users] Multiple sip proxy per * server.

2006-06-05 Thread Wai Wu





Anyone know how to 
direct sip calls in a dial plan to a specific proxy if * is registered with more 
than one proxy?
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Re: [Asterisk-Users] Can´t send emails