date:20050828

Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the coarse-training from scratch?
  
   Thanks Rich for the comments. Obviously you got my point. And I was
   hoping
   that it is possible somehow to do #1 (coarse-training from scratch) 
   after
   a
   PSTN line is transfered to another phone or something very important
   changes
   about a call. But considering how complicated a situation this might 
   be
   (there is music on hold to say the least, or perhaps the transferee 
   may
   reject the call and the call returns to the operator [in which case 
   you
   wouldn't want #1], etc.) I guess this is not a simple task. Anyway, 
   I'll
   keep on watching for a solution.
  
   The echo canceler preload happens shortly after the analog line is
   seized. Since a call transfer does not open/close the pstn line again,
   its not going to preload again. However, it should not have to anyway
   since there hasn't been any electrical changes there.
  
   What type of phones are you using internal when you're transferring a
   call?
 
  The internal phones involved were both analog phones, i.e. Zap channels.
  This problem happened once in the last 2 days (with KB1), so I'm not too
  negative.
 
  Are the analog phones connected to a TDM card, channel bank, or what?
 
  cvs head or stable  version?
 
 I have 2x TDM cards (4x FXS + 3x FXO), so all analog phones and pstn lines 
 are connected to TDM cards, Rev E/F.
 
 Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 
 2005-08-19 22:03:57 UTC
 
 That's on an Athlon 2000+, with CentOS 4.1, 2.6.11 kernel.
 
 (I have the monitor recording of that call, and I've listened to it a couple 
 of times. When the caller from pstn is talking to the operator [before 
 transfer] I can't hear echo, but it's a very short converstion, so now I 
 can't be sure %100. But after the transfer I can clearly hear the echo. I 
 should catch another case like this.)
 
 Thanks for the splitter answer, btw.

I don't use the fxs modules on a TDM card, so all I can do is guess
that the internal call handling (eg, drivers) for the fxs is suspect.

The guess is based on an assumption that *'s echo canceler is involved
with fxs modules since there _is_ a hybrid involved, and, another set
of canceler parameters are involved with the fxo side of the call.
(In effect, two instances of the canceler per fxo-fxs call.)

Since you've mentioned the echo after transfer doesn't happen on a
regular basis, that would suggest the problem is not a coding issue
as that would likely involve echo on every transfer. If its not a coding
issue, then its likely the issue is related to interaction of the two
echo canceler instances.

Might try playing around with the canceler parameters on the fxs channel.
Since the analog fxs phone is always very close physically, maybe play
with the echotraining (echocancel=32, and other echo parameters) to 
see what impact those might have. (In theory, using something like
echotraining=800 on the fxo port and echotraining=200 on the fxs port
might influence the interaction, if that really is the issue.)


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Re: [Asterisk-Dev] Re: [Asterisk-Users] Help Solving Asterisk Lockups

2005-08-28 Thread James Jones

If this issue exists doesn't it mean that asterisk is unstable anyway?

On Sat, 2005-08-27 at 16:29 -0400, Marc Olivier Chouinard wrote:

I have repeatedly mention this issues, and I keep getting laugh at from
Mark... So I do not think donation to digium will fix the core problem.

Digium want to sell the product like it is rightnow, and have no plan to
do masive change to fix any core problems. They think that if they
start redesign this, it will bring back asterisk to be unstable again.

Marc O.

James Jones wrote:

I know of good way to solve this problem. I have been authorize by my
company to try to a group of people and businesses to give donations
to get Digium to fix this issue. We will start the pot at $200. Are
there any takers?

On Sat, 2005-08-27 at 10:08 -0400, [EMAIL PROTECTED] wrote:

So the only thing we have in common is the remote monitoring ...

Are you using:

1) Realtime (and if so, with mysql, odbc, etc?)
2) Logging CDR records? (and if so, how)

This post looks like it could pertain to the same problem:
http://lists.digium.com/pipermail/asterisk-dev/2005-August/014797.html
.. but I don't think it has been resolved.

Eric

Julian
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Now I'm worried - we have exactly the same problem, but were going to
upgrade to 1.2. Now it seems as if CVS-HEAD has the same issue.

We have a TE405P, with 80 cisco7960 phones connected to a isdn30 pri.
The same issues ocurr - Busy on inbound calls, cannot place outbound,
nothing in the logs.

Are you (as we are)

1) running with queues and agents

We are *not* using queues or agents.

2) reloading the config (reload from the cli)

I have used restart now from the cli to bring the system back when it
freezes. Honestly I'm not sure that I've tried a plain reload. I'll
see if that brings it back next time it dies.

3) monitoring the system by connecting to the manager cli ?

We have an application (similar to the Flash Operator Panel) that
connects to the manager API (via port 5038, not the CLI) and is used by
our receptionist to monitor extensions and transfer calls.

I intend to slowly start stripping the system down. Next time it crashes
I will change the logging from mysql to csv only. This bug makes it
sound like an mysql glitch can cause the system to hang:
http://bugs.digium.com/view.php?id=4953

We are looking for all possible solutions to this.

Me too!

Eric

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Re: [Asterisk-Users] Need quote for Asterisk and billing remote install

2005-08-28 Thread Paul

Based on the fine detail you provided my estimate is somewhere between 1 
and 10 thousand US dollars.


Vikas wrote:


I need the quote please. Would appreciate a off list quote.

thanks

On 8/28/05, Chris Felter [EMAIL PROTECTED] wrote:
 


Please send me a quote for remote installation of
Asterisk, GUI administration, and billing for calling
card, caller ID based prepaid, and postpaid.

Off list please.

   



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[Asterisk-Users] Re Invite not working

2005-08-28 Thread Ishay






Hi 

Although canreinvite option is yes, 
the asterix doesn't send reinvite and the media is going through the asterix 
instead of between the two sip phones. 

Both sip phones (handytone 486) are 
configure with canreinvite option yes and use the same codec G.729. And 
Dial() 
command don't contains t or T.

Any suggestion on 
what could be the problem ?

Thanks,

Ishay 

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RE: [Asterisk-Users] no sound with red alarm?

2005-08-28 Thread Chad Osmond




I 
have had no issues where asterisk is affected by a Sangoma card being 
down.
I 
ran my test server like that for a few weeks doing lots of testing before I 
brought it up with a dummy card. Even now, if it's up or down it doesn't matter 
to asterisk.

Chad


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Damon 
EstepSent: August 25, 2005 10:20 PMTo: 
Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] no sound 
with red alarm?


Does it make sense that a system 
with a single PRI (sangoma card) would loose its ability to play sounds when not 
connected to the PSTN for clock?

Is there a way to configure ztdummy 
as a BACKUP clock source, or is there a better way 
altogether?
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

2005-08-28 Thread Andrew Kohlsmith

On Sunday 28 August 2005 10:21, Rich Adamson wrote:
 Might try playing around with the canceler parameters on the fxs channel.
 Since the analog fxs phone is always very close physically, maybe play
 with the echotraining (echocancel=32, and other echo parameters) to
 see what impact those might have. (In theory, using something like
 echotraining=800 on the fxo port and echotraining=200 on the fxs port
 might influence the interaction, if that really is the issue.)

I'd suggest turning off echotraining on the FXS altogether, and perhaps even 
killing the echocanceller on FXS entirely.  (you won't be getting significant 
echo from the FXS, and the FXO should be handling it anyway) -- 
echocancelwhenbridged might be an interesting thing to play with as well.

e.g. (assuming port 1-3 are FXO and port 4-7 are FXS)

echocancel=64
echocancelwhenbridged=yes
echotraining=800
channel = 1-3

echocancelwhenbridged=no
channel = 4-7

type of thing...  I'm just throwing out some ideas here and have not tried it 
myself.

-A.
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Re: [Asterisk-Users] Detect Dialtone

 i need to know something in the zaptel configuration
 as it seems i can configure detecting the busy tone and hangup after number 
 of busy tone 
counts, that was great but the problem is sometimes the pstn line has no 
dialtone and when i 
try to make call it continue dialing while not having a dialtone! while it 
should say all 
lines are busy/congested how can i configure that??
 
 i already done (immediate=no) and still it opens the zap trunk even when 
 theres no dialtone 
and shows that zap/3 answered
 
 -
 ;Specify whether the channel should be answered immediately or
 ; if the simple switch should provide dialtone, read digits, etc.
 ;
 immediate=no
 --

I might be way off base here, but the immediate=no parameter is oriented
towards incoming zap calls (not outgoing calls), and the callprogress
and busy detect stuff was intended to detect busy tones (not dial tone).
I don't think there is any logic in the zap channels to listen for
dial tone before dialing. (But, I could be wrong.)

What are you using for the zap fxo channel (eg, channel bank, tdm, x100p)?

Are you trying to play with the case where you have an analog phone
bridged on your fxo line, and detect the lack of dialtone when 
someone is using that analog phone?


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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

  Might try playing around with the canceler parameters on the fxs channel.
  Since the analog fxs phone is always very close physically, maybe play
  with the echotraining (echocancel=32, and other echo parameters) to
  see what impact those might have. (In theory, using something like
  echotraining=800 on the fxo port and echotraining=200 on the fxs port
  might influence the interaction, if that really is the issue.)
 
 I'd suggest turning off echotraining on the FXS altogether, and perhaps even 
 killing the echocanceller on FXS entirely.  (you won't be getting significant 
 echo from the FXS, and the FXO should be handling it anyway) -- 
 echocancelwhenbridged might be an interesting thing to play with as well.
 
 e.g. (assuming port 1-3 are FXO and port 4-7 are FXS)
 
 echocancel=64
 echocancelwhenbridged=yes
 echotraining=800
 channel = 1-3
 
 echocancelwhenbridged=no
 channel = 4-7
 
 type of thing...  I'm just throwing out some ideas here and have not tried it 
 myself.

That certainly makes more sence then my logic did. :)


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RE: [Asterisk-Users] no sound with red alarm?









Do you have any other zaptel hardware in
the machine?



Sangoma did confirm this was an issue that
was corrected in beta13 of the the wanpipe drivers.



Asterisk does require a timing source,
either a zaptel card or ztdummy to function correctly.













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Chad Osmond
Sent: Friday, August 26, 2005 9:07
AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] no
sound with red alarm?





I have had no issues where asterisk is affected by a Sangoma
card being down.

I ran my test server like that for a few weeks doing lots of
testing before I brought it up with a dummy card. Even now, if it's up or down
it doesn't matter to asterisk.



Chad









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: August 25, 2005 10:20 PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] no sound
with red alarm?

Does it make sense that a system with a single PRI (sangoma
card) would loose its ability to play sounds when not connected to the PSTN for
clock?



Is there a way to configure ztdummy as a BACKUP clock
source, or is there a better way altogether?








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RE: [Asterisk-Users] T1 DSU's/Split for voice

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Paul
 Sent: Sunday, August 28, 2005 7:48 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] T1 DSU's/Split for voice

 [EMAIL PROTECTED] wrote:

 Hello All,
 I was wondering if I could do the following on asterisk...

 Get a T1 between 2 locations, and split it into a data channel of
like
 1024, and use the rest for voice channels.

 Has anyone done this and had it working well?  Or would I need to get
a
 csu that allows a split into two interfaces?

 Regards,
 Greg

 On a point-to-point t you will need hardware to split channels at both
 ends.

 Since you posted to this list I assume that * is used at least on one
 end of the t1. If you use good routers that can prioritize voip
traffic
 then all you need is ata's and/or ip phones at the other end. That
gives
 you more data bandwidth when the phones are not in use. Segregating
n/24
 of your t is not so flexible.

That would be one way to do it, but both the Digium cards and the
sangoma cards can be configured to route some of the 24 timeslots on a
T1 to voice and some to data.
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[Asterisk-Users] Re: DIALSTATUS for Originate Command

2005-08-28 Thread Stefan Tichy

On Sun, Aug 28, 2005 at 10:45:18AM -, saket  setu wrote:
 I am trying to use the originate Command from the Asterisk manager on both 
 SIP and ZAP.
 The command works successfully but does not return any DIALSTATUS...

Response: Success
Message: Originate successfully queued

Indeed this response to a originate manager command is not what you
may have expected. You can listen to the events provided by the
manager interface and wait for something like this:

Event: Newstate
Channel: SIP/201-
State: Up


 I have also tried to write a dial plan in a manner such that i originate a 
 call
 to my internal extension and jump to a context in the dial plan and execute
 the Dial command and trap all the statuses but this also does not work and it
 straight away bridges my internal extension to the external call without 
 returning any dial status.

No dial plan application will be executed until someone picks up at
the sip phone. If the dial plan gets involved the sip channel has
been answered.


 Here is the example of what i did:
 1. Originate:
 Action: Originate
 Channel: SIP/201 (Internal extension)
 Context: Airtel
 Extension: 26191341(External PSTN Number)
 Priority: 1

 2. Dial Plan :
 [AIRTEL]
 exten = _XX.,1,Dial(ZAP/${EXTEN},15,t)
 exten = _XX.,2,NoOp(${DIALSTATUS})
 exten = _XX.,3,Goto(_XX.-${DIALSTATUS},1)

The target extension looks strange. But do not use
Goto(${EXTEN}-${DIALSTATUS}) here. _XX. will match this and you
get an endless loop.


-- 
Stefan Tichy   [EMAIL PROTECTED]
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Re: [Asterisk-Users] Low handset microphone volume with Sipura SPA-841

2005-08-28 Thread John Novack

I have not experienced that problem, but earlier firmware resulted in an 
unusable speakerphone.

Check if you have the latest firmware, then ask Sipura support for help.
The one time I E-mailed them they were quite responsive.

the 841 still has a worthless display though, doesn't it?
Lack of backlightimg and too small isn't going to be fixed by a firmware 
change!


John Novack


Juan Jose Comellas wrote:

I have just bought several Sipura SPA-841 SIP phones, and after some testing I 
have found out that the volume received by other parties when calling using 
the handset is very low. I've been able to reproduce this problem in the 3 
phones I've tested so far. I've tried tweaking several configuration options 
but nothing I has helped so far.


Has anybody else experienced this problem? There are only two holes for the 
microphone in the handset and they are really small. I was thinking that 
myabe this is the cause. Any thoughts?



 


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RE: [Asterisk-Users] How to use * and # as part of number in dial command


* # are valid in a dialplan

you would start your exten = with the vertical service code *21*
then play prompt, collect digits, play prompt, dial
${exten}$(var_for_collected_digits}

BUT, unless I have missed something, You can just send *21* to the PSTN
and then follow their prompts! As long as DTMF is configured correctly
it should work. If they don't prompt you still dial the same way,
activate the vertical service code with *21*

Exten *21*,1,Dial(ZAP/CHAN/{$EXTEN})

That's all! Then just enter the rest of the digits, allowing the PSTN
switch to collect the DTMF and activate the code.

You may want to use something like this to specify which ZAP channel
your forward (or which line).

Add a line number first like this

Exten 1*21*,1,Dial(ZAP/CH1/*21*)

Notice we don't send the 1, but we do use it to pick the zap channel.



From the wiki pages

http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.
conf

Extensions 
An extension can be one of two types: a literal or a pattern. 

A literal extension can be a number, like 123, and it can also contain
the standard symbols * and # that appear on ordinary telephones, so
12#89* is a valid extension. Some telephone keypads have the special
DTMF keys labeled A, B, C and D, and extensions can be defined with
these letters too. In fact, the name of a extension can contain any
letter or number as well as some punctuation marks. Note that many VOIP
telephones are able to dial extension numbers that may be any
arbitrary text string, such as Office. It is perfectly permissible to
define an extension with the name Office in Asterisk. 

Are extension names case sensitive? Well, they are and they aren't. They
are case sensitive in the sense that when Asterisk is trying to match
the extension a user dialed against the extensions defined for a
context, the extension must match, including case. So if a user dials
extension OFFICE using their VOIP telephone, Asterisk will not start
executing the commands you have defined for an extension named Office.
On the other hand, extension names are not case sensitive in the sense
that you can not define different extensions (in the one context) that
have the same names differing only in case. So you can't define one set
of commands for extension Office and another set of commands for
extension OFFICE.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Francesco Peeters
 Sent: Sunday, August 28, 2005 2:45 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] How to use * and # as part of number in
dial
 command
 
 On Sat, August 27, 2005 23:41, Michel Koenen said:
  Hi all,
 
  I am struggling with the following and I can't get it work:
 
  In the Netherlands where I live it is possible to use special codes
  (aka vertical service codes) to set special 'behaviour' of
phonecalls.
  So e.g. when I want to dial out with a normal phone and I dial
  *31*phonenumber to dial then it will turn off my numberindication
  (CID) at the called party.  They seem to call this the 'keypad
  protocol' but I  cannot find this term when searching through
asterisk
  documents.
 
  My asterisk system is connected to an ISDN line with HFC card. I use
  zaphfc module for that.  In my extensions I tried several things to
  dial out and use the *31* but without success.
 
 
 A few others are:
 Call forwarding: *21*number# / #21#
 Delayed forwarding: *61*number# / #21#
 Busy forwarding: *63*number# / #63#
 
 (the ## numbers are to disable the service)
 
 I am glad to see this topic come up before I ran in to it myself!  ;-)
I'm
 curious about the answers...
 
 
 --
 Francesco Peeters
 
 GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
 If your program doesn't recognize my signature, please visit
 http://www.CAcert.org/index.php?id=3 to retrieve the Root CA
certificate.
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Re: [Asterisk-Users] Detect Dialtone

2005-08-28 Thread John Novack




bodra wrote:


i need to know something in the zaptel configuration
as it seems i can configure detecting the busy tone and hangup after number of busy tone 
counts, that was great but the problem is sometimes the pstn line has no dialtone and 
when i try to make call it continue dialing while not having a dialtone! while it should 
say all lines are busy/congested how can i configure that??

 

As has been said many times before, there is NO dialtone detection on 
either the X100P or TDM400 FXO. IT was either never written, doesn't 
work, or has been disabled.
No one seems to be interested in correcting that situation, except for 
those of us who don't have the ability.
If someone smarter than I would determine if there is either a bug 
report or feature request, IMO this is FAR more important than some new 
whizz bang features in 1.2
Analog lines are, and will be a reality for some time to come, and needs 
to be addressed.
Inserting w into the dial string also seems to not work in HEAD, but 
does in 1.0.9, so misdialing persists.


Immediate=no or yes is for incoming calls.

John Novack


i already done (immediate=no) and still it opens the zap trunk even when theres no 
dialtone and shows that zap/3 answered

-
;Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
;
immediate=no
--

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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

 
  Then, on a commercial turn up (back when I did these, it was Western 
  Union and/or MCI), the tech at the other end would again dialup the 
  milliwatt, report the value measured over the loop and the pad(s) 
  re-adjusted to match the values for the loss in a document provided.
 
 That is the device called that would measure the milliwatt db loss when 
 plugged into an analog port and dialed into a milliwatt line?

Take a look at 
 http://www.repaircalibration.com/triplett-triplett-telco-testers.html

The model 2 is fine for simply measuring line loss, noise, etc.

The model 5 is the same but with a tone generator as well. (Handy 
for sending a tone from one ATA and measuring the received tone
on another ATA.)

There are other companies besides Triplett that make equivalent units.
The model numbers (eg, 2, 5) has its roots in the old Western Electric
specifications, and several manufactureres stuck with those.

Google terms: transmission test set, subscriber loop test set
(450,000 hits).

A simple analog Voltmeter will also work in some cases for measuring
telephony audio tones. If you can find one with a scale marked in db,
try it. Don't forget to place a 600 ohm resistor across tip  ring
as the cable pair _must_ be terminated (lot left wide open) to obtain
accurate values. (Use an old pots telephone to dial the CO milliwatt,
add the 600 ohm resister after you reached to milliwatt, hang up the
pots telephone, and measure the loss.)

Another approach is to have the telco come out on a low volume
trouble ticket and ask the technician what values he read to the
milliwatt generator. ;)


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RE: [Asterisk-Users] SIP Benchmarking / Stress Testing

2005-08-28 Thread Alex Vishnev









sipsak (www.sipsak.org. ) is an excellent tool for
this.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Friday, August 26, 2005
10:48 AM
To: 'Asterisk Developers Mailing
List'
Cc: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] SIP
Benchmarking / Stress Testing







Anyone have a good tool(s) to use for simulating a bunch of
calls? Benchmarking or stress testing?











I only need SIP protocol, and do appreciate any replies...I
realize I could google it, but I am looking for opinions as well.









Sherwood McGowan






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Re: [Asterisk-Users] WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type

2005-08-28 Thread Eric Wieling aka ManxPower


Giorgio Incantalupo wrote:

Hi,
is there anybody who knows what this warning means??

WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type


I would bet that [10] doesn't have a type=
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Re: [Asterisk-Users] ignorepat not working - what might I have done?

2005-08-28 Thread Mason Loring Bliss

On Fri, Aug 26, 2005 at 12:31:29PM -0500, Eric Wieling aka ManxPower wrote:

 ignorepat does not work for SIP since the dialtone is coming from the 
 SIP device, not from Asterisk.
 
 You would need to set the phone up to continue dialtone after dialing 9. 
  Not all phones support that.

Hm. In this case, I suspect that the dial tone isn't coming from the
phone... I have two possible places to route calls starting with 9,
and only one of them is a SIP device... In fact, I just commented out
the pattern that leads to the SIP device, so the only thing left is
a transfer into a voice menu, and I still get no dial tone after the
leading 9...

Continuing thanks in advance for more ideas... :)

-- 
 Mason Loring Bliss [EMAIL PROTECTED]http://blisses.org/  
I am a brother of jackals, and a companion of ostriches.  (Job 30 : 29)
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Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards

2005-08-28 Thread Philipp von Klitzing

Hi!

 We are in the process of an Asterisk call center deployment using IAX2
 G711 ulaw softphones.   Outbound sound quality is terrible.  

Have you tried a different sound card and/or a USB handset (which 
includes an external sound card)? And what exactly do you mean with 
terrible sound?

Philipp


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[Asterisk-Users] motorola vt1000 games

2005-08-28 Thread trixter http://www.0xdecafbad.com

For those that are interested in the vt1000 paper I wrote a while back,
I have it now on my webpage, at
http://www.0xdecafbad.com/Unlocking-Motorola-VT1000.html

Some of the information there was posted elsewhere, some wasnt.
basically the unit runs vxworks, and it needs a docsis like server to
reconfigure properly (its different from the cable modem docsis stuff
mot does) but ...

I think most if not all the hardware is supported in linux, I didnt
really check just glanced at the chips, it may be possible to reflash
the unit with an embedded linux version and run a very stripped down
asterisk implementation, thus making the units more valuable, and since
there are many in surplus now with no large provider supporting them
anymore, you may be able to get em really cheap (I believe ebay has some
for cheap).

There is support to reflash without difficulty, providing you use the
vxworks boot loader, there is a connector that may be jtag which would
let you more easily reflash without doing dev work under vxworks to
write a loader app on the unit.  Again this is stuff I didnt really look
into.  Downside is that JTAG is slow so you prolly wouldnt want to
reflash the whole thing via JTAG but just enough to get to the serial
port and read the rest of the data that way.  




-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] Sip pickup

2005-08-28 Thread Andrzej Nowrot



Hi,

  In my office I%u2019m using mixed architecture of Zap and Sip phones, 
everything works fine but I have got some problems with picking up Sip 
channels. To be certain I can%u2019t do it at all, after I%u2019m dialing *8 
the console says nothing to pick up (despite I configure appropriate 
callgroup and pickupgroup).

Do I need some additional application or Asterisk code modification to pickup 
Sip channel, if so, what should I install or modify?? Where can I find some 
guide or tutorial about this??
I was wondering of using AGI to solve my problem but I do not know if I achieve 
a goal :)

Please help ;)

Cheers


--
Startuj z INTERIA.PL!  http://link.interia.pl/f186c

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Re: [Asterisk-Users] 24 line softphone

2005-08-28 Thread Ben Brown





Here is my situation. I have
MeetMe conferences going on between internal SIP lines and Zap
channels. I need to be able to join each conference at the beginning
and end, and easily switch between them on request for monitoring. I
also need the option of joining the conference if needed. I have
achieves the effect I want using the xlite client and joining 3
conferences, placing each on hold and switching between them. I mute
the mic when monitoring. This leaves me the option of joining any
conference at the push of a button as needed.

I suppose if there was just a way to monitor the 24 conferences on
request, then the participation could be accomplished using a regular
SIP client.

Thanks

BEN

Mark Edwards wrote:

  Or is it a monitoring application that you need? for instance,  do you
need the ability to monitor active channels on request? The
description below isn't clear around what you mean in regard to
'monitorin' and 'placing the others on hold'. Normally you 'place
someone on hold' after you have spoken to them - so I guess I am not
clear on a few points.

Mark

On 8/28/05, Ben Brown [EMAIL PROTECTED] wrote:
  
  
I am looking for a single soft phone application that is capable of a
minimum of 24 concurrent lines. Suffice to say that I have a somewhat
unique application here, and I would like all connections active all the
time. I want to be able to switch between them for monitoring purposes,
placing the others on hold.

Thanks

BEN
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Re: [Asterisk-Users] 24 line softphone


Ben Brown wrote:

I suppose if there was just a way to monitor the 24 conferences on request, then 
the participation could be accomplished using a regular SIP client.


In CVS HEAD (and soon Asterisk 1.2), app_chanspy will do exactly that.
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Re: [Asterisk-Users] T1 DSU's/Split for voice

2005-08-28 Thread Paul


Damon Estep wrote:

 


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul
Sent: Sunday, August 28, 2005 7:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 DSU's/Split for voice

[EMAIL PROTECTED] wrote:

   


Hello All,
I was wondering if I could do the following on asterisk...

Get a T1 between 2 locations, and split it into a data channel of
 


like
 


1024, and use the rest for voice channels.

Has anyone done this and had it working well?  Or would I need to get
 


a
 


csu that allows a split into two interfaces?

Regards,
Greg


 


On a point-to-point t you will need hardware to split channels at both
ends.

Since you posted to this list I assume that * is used at least on one
end of the t1. If you use good routers that can prioritize voip
   


traffic
 


then all you need is ata's and/or ip phones at the other end. That
   


gives
 


you more data bandwidth when the phones are not in use. Segregating
   


n/24
 


of your t is not so flexible.

   


That would be one way to do it, but both the Digium cards and the
sangoma cards can be configured to route some of the 24 timeslots on a
T1 to voice and some to data.
 

That means having a pc with one of those cards at each end and knowing 
how to also config it as a router. I get standalone adtran t-1 routers 
for about $650 in qty 1.


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RE: [Asterisk-Users] 911 Notices

2005-08-28 Thread Dean Collins

Packet8 got around this in an interesting waycharge clients $1.50
per month for E911 or have the option of saying no.

Lol, how many people do you think took them up on that offer?


Dean


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of BJ Weschke
 Sent: Friday, 26 August 2005 6:24 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] 911 Notices
 
 On 8/26/05, Mark Phillips [EMAIL PROTECTED] wrote:
  Broadvoice sent out a notice and threatened to disconnect me if I
did
  not respond. If I disagreed with their stand they would disconnect
me
 too.
 
  I think they said something like we don't have it and we ain't
getting
  it. Click here to acknowledge.
 
  I'm guessing that the statement gets them off the hook?
 
 
 
  The way I understand it. Yes, for now.  That only allows them to
 be compliant  up until the mandatory compliance date. After that date
 passes, technically, you're supposed to offer it if you're business is
 interconnecting voip networks to the PSTN.
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

2005-08-28 Thread Steve Underwood


Matt Fredrickson wrote:


On Fri, Aug 26, 2005 at 02:00:54PM -0600, Rich Adamson wrote:
 


Relative to the fxotune app, it would appear the app is specific
to the v2.4 kernels (/dev/zap*), which the v2.6 kernels don't use
   



It should with 2.4 and 2.6.  2.6 kernels with properly configured udev
rules should create the /dev/zap/* entries dynamically.

 


(but rather the udev equivalent). (When I had * running on a v2.4
kernel, the output from fxotune never deviated from all zero's. So
I'm assuming the default chipset values were already tweaked by the
chipset manufacturer to US telco lines. If that is true, then 
running fxotune in the US has very little value.)
   



Sometimes in the US you still deal with line impedance issues.  In fact,
I was told by an engineer that worked for the company that designs the
line interface part that the bulk of echo problems (with line interface 
parts such as this) are related to AC impedance mismatches (which is one

reason why I haven't done the digital hybrid tuning portion of fxotune
still).  It should work the same regardless of which kernel (2.4 or 2.6)
you are using.
 

Everyone has to deal with line impedance issues. The hyrbids in 
exchanges usually use a compromise impedance, but there are tolerance in 
the circuits. The lines themselves vary far more, especially if loading 
coil, and other fudges have been used. To get pretty good hybrid 
performance you always need to individually tune.


What you have to remember about hybrids is they are not there to give 
great rejection. They are there to give enough rejection to prevent 
howling. Nothing more. The telephone approvals specs in most places only 
call for 12dB of suppression through a hybrid. The exchange specs tend 
to use a slightly higher figure, but only slightly.


I don't follow why knowing that impedance mismatch is the problem has 
stopped you making fxotune fix it. :-\ Where you the one who asked me 
how to make fxotune work well on IRC? Someone asked a while ago, and 
said they were working on a faster tuning algorithm for fxotune. I've 
forgotten who.



If it doesn't, and you have udev setup correctly, something is fundamentally
wrong in the setup.
 


Regards,
Steve

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Re: [Asterisk-Users] Sip pickup

   In my office I%u2019m using mixed architecture of Zap and Sip phones, 
 everything works 
fine but I have got some problems with picking up Sip channels. To be certain I 
can%u2019t do 
it at all, after I%u2019m dialing *8 the console says nothing to pick up 
(despite I 
configure appropriate callgroup and pickupgroup).
 
 Do I need some additional application or Asterisk code modification to pickup 
 Sip channel, 
if so, what should I install or modify?? Where can I find some guide or 
tutorial about this??
 I was wondering of using AGI to solve my problem but I do not know if I 
 achieve a goal :)
 
 Please help ;)

Unless I'm misunderstanding your request, it works just fine for me
on any sip extension using cvs-head from a couple of days ago.

Example: 
[3111]  
type=friend
username=3111
secret=mysecret
host=dynamic
context=from-sip
dtmfmode=rfc2833
callgroup=2
pickupgroup=2
mailbox=3111

When the above sip phone is ringing, on another sip phone (C7960) 
I press *8# and the above phone stops ringing, and I'm talking to
the person that called 3111.


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Re: [Asterisk-Users] 24 line softphone

2005-08-28 Thread Steve Edwards


On Sun, 28 Aug 2005, Kevin P. Fleming wrote:


Ben Brown wrote:

I suppose if there was just a way to monitor the 24 conferences on request, 
then the participation could be accomplished using a regular SIP client.


In CVS HEAD (and soon Asterisk 1.2), app_chanspy will do exactly that.


Chanspy will not let you join a conference, only spy.

I do something similar with this snippet.

[supervisor]
; monitor an agent
exten = s,1,background(please-enter-a-conference)
exten = _xx#,1, meetme(${EXTEN:0:6},ampqs)
exten = _xx#,n, goto(${CONTEXT},s,1)

You enter a conference number and #, to join with mute on. You can
*1 to unmute if you want to speak. Press # to exit the conference
and you are prompted to enter another conference number.

I've hacked on app_meetme a bit to control who hears enter and leave
sounds and the application exit code as well as let the supervisor
kick all users or admins.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline   [EMAIL PROTECTED]Fax: +1-760-731-3000
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Re: [Asterisk-Users] 911 Notices

2005-08-28 Thread Julio Arruda


Remarks inline

Dean Collins wrote:


Packet8 got around this in an interesting waycharge clients $1.50
per month for E911 or have the option of saying no.

Lol, how many people do you think took them up on that offer?
 

From what I understand, Packet8 had this option for quite some time. I 
used (more than one year ago) to be Packet8 customer.

I still use a couple of DTA310 in my * system :-)


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Friday, 26 August 2005 6:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 911 Notices

On 8/26/05, Mark Phillips [EMAIL PROTECTED] wrote:
   


Broadvoice sent out a notice and threatened to disconnect me if I
 


did
 


not respond. If I disagreed with their stand they would disconnect
 


me
 


too.
   


I think they said something like we don't have it and we ain't
 


getting
 


it. Click here to acknowledge.

I'm guessing that the statement gets them off the hook?


 


The way I understand it. Yes, for now.  That only allows them to
be compliant  up until the mandatory compliance date. After that date
passes, technically, you're supposed to offer it if you're business is
interconnecting voip networks to the PSTN.
___
   


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Re: [Asterisk-Users] storing voice messages in DB SQL

2005-08-28 Thread Matthew Boehm

Its the same syntax for every other config. Just look at every other config
option and replicate.

Odbctable=mytablename

Or

Odbctable = mytablename

-Matthew


 From: harry gaillac [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Sun, 28 Aug 2005 12:11:07 +0200 (CEST)
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] storing voice messages in DB SQL
 
 hello,
 
 According to docs/README.odbcstorage how can we set :
 
 ///
 The database name (from /etc/asterisk/res_odbc.conf)
 is in the
 odbcstorage variable in the general section of
 voicemail.conf.
 You may modify the voicemessages table name by using
 odbctable=??? in voicemail.conf
 ///
 
 what's the right syntax in voicemail.conf ?
 Harry
 --- Matthew Boehm [EMAIL PROTECTED] a écrit :
 
 Yes. Look in the apps/Makefile for USE_ODBC_STORAGE
 and read in the docs/
 for a table structure.
 
 Right now it is ODBC only.
 
 -Matthew
 
 From: harry gaillac [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List -
 Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Sat, 27 Aug 2005 16:12:09 +0200 (CEST)
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] storing voice messages
 in DB SQL
 
 Hello,
 
 Can we store voice messages in a database instead
 of
 files.
 
 Regards
 
 
 
 
 
 
 
 
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RE: [Asterisk-Users] T1 DSU's/Split for voice

 
 Hello All,
 I was wondering if I could do the following on asterisk...
 
 Get a T1 between 2 locations, and split it into a data channel of
 
 
 like
 
 
 1024, and use the rest for voice channels.
 
 Has anyone done this and had it working well?  Or would I need to
get
 
 
 a
 
 
 csu that allows a split into two interfaces?
 
 Regards,
 Greg
 
 
 
 
 On a point-to-point t you will need hardware to split channels at
both
 ends.
 
 Since you posted to this list I assume that * is used at least on
one
 end of the t1. If you use good routers that can prioritize voip
 
 
 traffic
 
 
 then all you need is ata's and/or ip phones at the other end. That
 
 
 gives
 
 
 you more data bandwidth when the phones are not in use. Segregating
 
 
 n/24
 
 
 of your t is not so flexible.
 
 
 
 That would be one way to do it, but both the Digium cards and the
 sangoma cards can be configured to route some of the 24 timeslots on
a
 T1 to voice and some to data.
 
 
 That means having a pc with one of those cards at each end and knowing
 how to also config it as a router. I get standalone adtran t-1 routers
 for about $650 in qty 1.
 
No doubt about that, not sure why the Greg wants to do what he is asking
for, personally I would use 2 used Cisco routers with integrated T1 DSUs
and priority queuing for VoIP, assuming VoIP phones at the remote
location. That would be about $600 for qty 2 on ebay!

I just assumed Greg had a need for a T1 TDM interface on both ends. 
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Re: [Asterisk-Users] 24 line softphone

2005-08-28 Thread Ben Brown





I have no problem joining the
conferences and monitoring. What I need is a nice, simple, preferably
GUI method to switch between multiple active connections. I have a
method I like using a 3 line softphone, which works for 3 conferences,
but I need one "line" for each connection to use my method, thus I am
seeking a 24 line softphone. 

Thanks for the thoughts

BEN

Steve Edwards wrote:
On Sun, 28 Aug 2005, Kevin P. Fleming wrote:
  
  
  Ben Brown wrote:


I suppose if there was just a way to
monitor the 24 conferences on request, then the participation could be
accomplished using a regular SIP client.
  


In CVS HEAD (and soon Asterisk 1.2), app_chanspy will do exactly that.

  
  
Chanspy will not let you join a conference, only spy.
  
  
I do something similar with this snippet.
  
  
[supervisor]
  
; monitor an agent
  
 exten = s,1,
background(please-enter-a-conference)
  
 exten = _xx#,1, meetme(${EXTEN:0:6},ampqs)
  
 exten = _xx#,n, goto(${CONTEXT},s,1)
  
  
You enter a conference number and "#," to join with mute on. You can
  
"*1" to unmute if you want to speak. Press "#" to exit the conference
  
and you are prompted to enter another conference number.
  
  
I've hacked on app_meetme a bit to control who hears enter and leave
  
sounds and the application exit code as well as let the supervisor
  
kick all users or admins.
  
  
Thanks in advance,
  

  
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867
PST
  
Newline [EMAIL PROTECTED] Fax:
+1-760-731-3000
  
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[Asterisk-Users] Multiple PCI cards

2005-08-28 Thread Garth van Sittert


Hi All

Does anyone know if multiple Digium cards on a single machine will be a 
problem.


Machine specs:  Dual Zeon 3.0GHz on Intel server board.
Cards:   TE411P, TDM400P, TDM400P

I will turn off all unnecessary PCI devices; USB, parallel, serial, etc...

Thanks
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RE: [Asterisk-Users] Multiple PCI cards

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Garth van Sittert
 Sent: Sunday, August 28, 2005 11:46 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Multiple PCI cards

 Hi All

 Does anyone know if multiple Digium cards on a single machine will be
a
 problem.

 Machine specs:  Dual Zeon 3.0GHz on Intel server board.
 Cards:   TE411P, TDM400P, TDM400P

 I will turn off all unnecessary PCI devices; USB, parallel, serial,
etc...

 Thanks

Have not tried it since November 2004, but at that time I ended up
replacing the FXO/FXS cards with sipura SPA3000 ?(check model number,
its been awhile). Each one gave 1FXO/1FXS port so 4 of them replaced 2 4
port TDM cads. Works well.

Again, this was almost a year ago, so look for more feedback for users
that have tried it with current hardware/firmware/software.
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[Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd

2005-08-28 Thread cmisip

There is a patch to mplayer that allows it to suppress stdout messages
and instead output pcm data to stdout.  I managed to get it working with
app_mp3.c and seems like it is working fine.  All that was needed was a
change in the execl line and a slight increase in timeout value.  I have
only done limited testing.  If mplayer is able to replace mpg123 without
issues, this opens up a whole lot of media that can be piped through
Asterisk.


I want to be able to send a dtmf key to asterisk and have mplayer
forward or rewind.

I dont know much about all this stuff ( I really am not a programmer )
but I have a lot of interest in it.  I use asterisk to play podcasts on
my cell phone (podcasts on the road) and the forward rewind feature
would be nice.  

Mplayer has a -slave command that allows it to read commands from stdin
I think. It also has the -input parameter which allows it to read
commands from a fifo.  I could get the -input  to work on the command
line sending commands to the fifo with bash echo.  However, when I am
actually listening to a podcast from MP3Player in asterisk, sending a
command to the fifo would cause MP3Player to respond to the command,
continue playback for a short time and then exit with a No More MP3
message and a bunch of Dropping voice to exceptionally long queue

Any ideas on how a forward, rewind feature can be implemented on
MP3Player cmd?

I have successfully used a bash script to interact with asterisk agi.  

My idea is to modify the app_mp3.c program to allow for a second
parameter to MP3Player to pass file position value.  Have app_mp3.c
store the value of f-subclass into res and exit with this value
whenever a key is pressed.  Bash could read this value and restart the
MP3Player application with a computed fileposition value as second
parameter.

This should achieve the result I want.  Is there a simpler way?

Any help or insight is appreciated.

Thanks.



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[Asterisk-Users] way to prevent voicemail dialout/callback from 'outside'

I am trying to find a way to allow dialout from voicemail when connected
from an 'internal' extension context, but prevent dialout when connected
from an 'external' extension context.

As far as I can tell the dialout context that can be set in voicemail
has no regard for the context from which the call to voicemail came in.

Any ideas on this? Maybe a variable passed when voicemail is called?
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RE: [Asterisk-Users] Multiple PCI cards

2005-08-28 Thread Asterisk

I have 2 TE410P's and a TDM400P in same machine without issues

Bart

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Damon Estep
Sent: Sunday, August 28, 2005 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Multiple PCI cards




 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Garth van Sittert
 Sent: Sunday, August 28, 2005 11:46 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Multiple PCI cards
 
 Hi All
 
 Does anyone know if multiple Digium cards on a single machine will be
a
 problem.
 
 Machine specs:  Dual Zeon 3.0GHz on Intel server board.
 Cards:   TE411P, TDM400P, TDM400P
 
 I will turn off all unnecessary PCI devices; USB, parallel, serial,
etc...
 
 Thanks


Have not tried it since November 2004, but at that time I ended up
replacing the FXO/FXS cards with sipura SPA3000 ?(check model number,
its been awhile). Each one gave 1FXO/1FXS port so 4 of them replaced 2 4
port TDM cads. Works well.

Again, this was almost a year ago, so look for more feedback for users
that have tried it with current hardware/firmware/software.
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Re: [Asterisk-Users] required packages for asterisk on FC3/FC4

2005-08-28 Thread Tzafrir Cohen

On Sat, Aug 27, 2005 at 12:13:43PM -0600, Damon Estep wrote:
 Can anyone shed some light on which of these packages are required and
 what component requires them? I am in the habit of putting them on, but
 in a few cases am not sure if they are still (or were ever) needed.
 
  
 
 qt-devel

huh?

 
 rpm-build
 
 gcc
 
 gcc-c++
 
 redhat-rpm-config
 
 gtk2-devel

Again: anybody actually using the built-in GUIs of asterisk?

 
 y ncurses-devel
 
 readline-devel
 
 bison
 
 krb5-devel
 
 openssl-devel
 
 cvs

Is it required for the build process itself? I don't think it belongs in
a BuildDepends line.

 
 patchutils
 
 libidn-devel

This one is a dependency of libcurl-devel (unless you use some buggy
version of some distro and have not bothered installing updates). No
need to install it explicitly.

 
 kernel-smp-devel (smp machine)
 
 flex (for a particular brand of pri card)

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Multiple IP's (aliases) on asterisk box?


Anyone have any experience running an asterisk box with a single nic
and multiple IP's (aliases)?

Have a six class-c production network that needs to be completely
re-IP'ed and need to run the box with both an old and new IP for a few
days.


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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

2005-08-28 Thread Andrew Kohlsmith

On Sunday 28 August 2005 11:59, Steve Underwood wrote:
 I don't follow why knowing that impedance mismatch is the problem has
 stopped you making fxotune fix it. :-\ Where you the one who asked me
 how to make fxotune work well on IRC? Someone asked a while ago, and
 said they were working on a faster tuning algorithm for fxotune. I've
 forgotten who.

I thought fxotune set up the built-in FIR filter in the DAA and nothing more.  
I'm really uncertain how a little filter is going to help with impedance 
matching, as it's still the same frequency ranges that need to get through to 
be digitized.

I have, however, been known to be mistaken on more than one occassion.  :-)

-A.

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Re: [Asterisk-Users] Multiple IP's (aliases) on asterisk box?

2005-08-28 Thread Bohuslav Coufal

Thats works without any problems.

Bob.

Dne neděle 28 srpen 2005 21:46 Rich Adamson napsal(a):
 Anyone have any experience running an asterisk box with a single nic
 and multiple IP's (aliases)?

 Have a six class-c production network that needs to be completely
 re-IP'ed and need to run the box with both an old and new IP for a few
 days.


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Re: [Asterisk-Users] Multiple IP's (aliases) on asterisk box?

2005-08-28 Thread Doug Lytle


Rich Adamson wrote:


Anyone have any experience running an asterisk box with a single nic
and multiple IP's (aliases)?

Have a six class-c production network that needs to be completely
re-IP'ed and need to run the box with both an old and new IP for a few
days.

 



I'm doing this with just 2 IPs, and not having any issues.  Phones on 
both the 10.10.10.0 and 192.168.102.0 networks.


Doug

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RE: [Asterisk-Users] error compiling on solaris 10

2005-08-28 Thread Frank Tarczynski


Message: 11
Date: Sun, 28 Aug 2005 11:46:29 +0800
From: chris [EMAIL PROTECTED]
Subject: [Asterisk-Users] error compiling on solaris 10
To: Asterisk Users Mailing List - Non-Commercial Discussion

asterisk-users@lists.digium.com

Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1



hello,



i change my OS from solaris 9 to solaris 10, tried running make to install 
asterisk but i'm getting the error below:



make -C editline libedit.a


To start try using gmake.  It's there, just add it to your PATH.

Frank



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[Asterisk-Users] 1.2.0 Beta1

2005-08-28 Thread Anton Krall

Guys. I was checking the changes for 1.2.0 beta1 and I read this:

* Asterisk Realtime Architecture
* Asterisk Manager Interface
* Asterisk Extension Language
* Dialplan functions
* More powerful dialplan expression parser
* Portability enhancements for FreeBSD, OpenBSD, Solaris and Mac OS X
* ... and many more!

Can somebody explain a bit mor ethe part regarding Dialplan fnuctions and
the Manager Interface? What are the changes?

Also, maybe it was me but I upgraded on a test server by doing the make and
make install over my cvs old one and when on the CLI I do a show version and
I get this:

server2*CLI show version
Asterisk  built by [EMAIL PROTECTED] on a i686 running Linux on
2005-08-27 22:54:47 UTC
server2*CLI

No version showing, is it just me?

Also checked the show applications and saw no new ones there.. Did I
messed up upgrading over the old one?

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Re: [Asterisk-Users] DIALSTATUS for Originate

2005-08-28 Thread Geoff Karl

On 28 Aug 2005 10:35:34 -, saket  setu [EMAIL PROTECTED] wrote:
 
 

  Hi all,
  I am from India and has been recently using asterisk for testing and 
 enahncing my telephony knowledge. I am trying to use the originate Command 
 from the Asterisk manager on both SIP and ZAP. The command works successfully 
 but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER as in case of 
 command DIAL when used from the dial plan. Can some one guide me how to get 
 the vaue of $DIALSTUATUS on originate or is there some other way to trap the 
 status both on SIP and ZAP.
  
  I have also tried to write a dial plan in a manner such that i originate a 
 call to my internal extension and jump to a context in the dial plan and 
 execute the Dial command and trap all the statuses but this also does not 
 work and it straight away bridges my internal extension to the external call 
 without returning any dial status.
  
  Here is the example of what i did:
  1. Originate:
  Action: Originate
  Channel: SIP/201 (Internal extension)
  Context: Airtel
  Extension: 26191341(External PSTN Number)
  Priority: 1
  
  2. Dial Plan :
  [AIRTEL]
  exten = _XX.,1,Dial(SIP/${ETEN},15,t)
  exten = _XX.,2,NoOp(${DIALSTATUS})
  exten = _XX.,3,Goto(_XX.-${DIALSTATUS},1)
  exten = _XX.-Busy,1,Hangup
  exten = _XX.-NOANSWER,1,Hangup
  exten = _XX.-ANSWER,1,Goto(s,1)
  exten = s,1,Queue(Airtel|r|||300)
  
  thanks
  Saket 

Stefan Tichy Wrote:

Response: Success
Message: Originate successfully queued

Indeed this response to a originate manager command is not what you
may have expected. You can listen to the events provided by the
manager interface and wait for something like this:

Event: Newstate
Channel: SIP/201-
State: Up

--


If you are using Async and the action ID for some reason the Event:
Newstate doesn't respond with the ActionID, but only a automatically
generated Uniqueid.

Any ideas on how to determine which ActionID is being returned?

Thanks,

Geoff
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[Asterisk-Users] Good Deal on A Good Asterisk Box?

2005-08-28 Thread asterisk




Go here http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=pe1850-mins=bsd

You can get dual xeon 3Ghz processors and then one 
of three other optional upgrades for free.
1 GB RAM or Free Embedded RAID, or free 73 Gig 
10KSCSI second HD (comes with one)

From the Digium site:
The Dell PowerEdge 1850 has an onboard Intel e1000 Ethernet controller that 
can interfere with the operation of the TE411P, TE410P, TE406P and TE405P, 
TE210P, TE205P cards. The recommended action for this server is to disable the 
onboard Ethernet controller and use a PCI-based solution. 

I am about to buy this for a total of $1,220 
including shipping.

Any comments? I think its a great box for a 
two te410p card setup. What do you think

Thanks,
Steve

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[Asterisk-Users] ztdummy and Linux 2.6.13-rc7

2005-08-28 Thread Doug Lytle

Anybody having issues with ztdummy under the current 2.6 RC7?  I get the 
following errors when trying to modprobe ztdummy:


Unable to register zaptel rtc driver

Doing a Google on the error shows reference to a message from 2004 that 
said you might not have RTC compiled into the kernel.  Checking via:


cd /usr/src/linux-2.6.13-rc7
grep -i rtc .config

shows:

CONFIG_APM_RTC_IS_GMT=y
CONFIG_RTC=m
CONFIG_GEN_RTC=m
CONFIG_GEN_RTC_X=y
CONFIG_HPET_RTC_IRQ=y
CONFIG_SENSORS_RTC8564=m
CONFIG_SND_RTCTIMER=m


Any suggestions?

Doug



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[Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7

2005-08-28 Thread Doug Lytle


Doug Lytle wrote:

Anybody having issues with ztdummy under the current 2.6 RC7?  I get 
the following errors when trying to modprobe ztdummy:



Failed to mention that this was under the current Asterisk 1.2 Beta 1 
release


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RE: [Asterisk-Users] Good Deal on A Good Asterisk Box?









How many PCI slots? You have to add a PCI
NIC and use 1 of them !!!













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of asterisk
Sent: Sunday, August 28, 2005 5:52
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Good
Deal on A Good Asterisk Box?







Go here http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=pe1850-mins=bsd











You can get dual xeon 3Ghz processors and then one of three
other optional upgrades for free.





1 GB RAM or Free Embedded RAID, or free 73 Gig 10KSCSI
second HD (comes with one)











From the Digium site:





The Dell PowerEdge 1850 has an onboard Intel e1000 Ethernet controller
that can interfere with the operation of the TE411P, TE410P, TE406P and TE405P,
TE210P, TE205P cards. The recommended action for this server is to disable the
onboard Ethernet controller and use a PCI-based solution. 











I am about to buy this for a total of $1,220 including
shipping.











Any comments? I think its a great box for a two te410p
card setup. What do you think











Thanks,





Steve
















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[Asterisk-Users] bid on this small project if you are interested.

2005-08-28 Thread john mills

http://www.rentacoder.com/RentACoder/misc/BidRequests/ShowBidRequest.asp?lngBidRequestId=325748




Start your day with Yahoo! - make it your home page 
http://www.yahoo.com/r/hs 
 
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Re: [Asterisk-Users] Good Deal on A Good Asterisk Box?

2005-08-28 Thread asterisk




I am assuming two, couldn't a USB NIC be 
used? Obviously not gigabit but can anyone see any problems with that 
setup?


  - Original Message - 
  From: 
  Damon Estep 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Sunday, August 28, 2005 4:32 
  PM
  Subject: RE: [Asterisk-Users] Good Deal 
  on A Good Asterisk Box?
  
  
  How many PCI slots? 
  You have to add a PCI NIC and use 1 of them 
  !!!
  
  
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Sunday, August 28, 2005 5:52 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: [Asterisk-Users] Good Deal on A 
  Good Asterisk Box?
  
  
  Go here http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=pe1850-mins=bsd
  
  
  
  You can get dual xeon 3Ghz 
  processors and then one of three other optional upgrades for 
  free.
  
  1 GB RAM or Free Embedded RAID, or 
  free 73 Gig 10KSCSI second HD (comes with 
  one)
  
  
  
  From the Digium 
  site:
  
  The Dell PowerEdge 1850 has an onboard Intel e1000 
  Ethernet controller that can interfere with the operation of the TE411P, 
  TE410P, TE406P and TE405P, TE210P, TE205P cards. The recommended action for 
  this server is to disable the onboard Ethernet controller and use a PCI-based 
  solution. 
  
  
  
  I am about to buy this for a total 
  of $1,220 including shipping.
  
  
  
  Any comments? I think its a 
  great box for a two te410p card setup. What do you 
  think
  
  
  
  Thanks,
  
  Steve
  
  
  
  

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Re: [Asterisk-Users] Multiple PCI cards

2005-08-28 Thread asterisk

But how many SIP-Zap channels can you have in use simultaneosly?  It it a
case of the usage?  Does the fact that the machine can handle these devices
also mean that it can use them at the same time, depending on the CPU and
RAM?


- Original Message - 
From: Asterisk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 28, 2005 2:23 PM
Subject: RE: [Asterisk-Users] Multiple PCI cards


 I have 2 TE410P's and a TDM400P in same machine without issues

 Bart

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Damon Estep
 Sent: Sunday, August 28, 2005 10:59 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Multiple PCI cards




  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Garth van Sittert
  Sent: Sunday, August 28, 2005 11:46 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Multiple PCI cards
 
  Hi All
 
  Does anyone know if multiple Digium cards on a single machine will be
 a
  problem.
 
  Machine specs:  Dual Zeon 3.0GHz on Intel server board.
  Cards:   TE411P, TDM400P, TDM400P
 
  I will turn off all unnecessary PCI devices; USB, parallel, serial,
 etc...
 
  Thanks


 Have not tried it since November 2004, but at that time I ended up
 replacing the FXO/FXS cards with sipura SPA3000 ?(check model number,
 its been awhile). Each one gave 1FXO/1FXS port so 4 of them replaced 2 4
 port TDM cads. Works well.

 Again, this was almost a year ago, so look for more feedback for users
 that have tried it with current hardware/firmware/software.
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Re: [Asterisk-Users] Low handset microphone volume with Sipura SPA-841

2005-08-28 Thread Rob Lith

In Admin/Advanced have you tried the Handset Input Gain: settings?
Rob

On 8/28/05, Juan Jose Comellas [EMAIL PROTECTED] wrote:
 I have just bought several Sipura SPA-841 SIP phones, and after some testing I
 have found out that the volume received by other parties when calling using
 the handset is very low. I've been able to reproduce this problem in the 3
 phones I've tested so far. I've tried tweaking several configuration options
 but nothing I has helped so far.
 
 Has anybody else experienced this problem? There are only two holes for the
 microphone in the handset and they are really small. I was thinking that
 myabe this is the cause. Any thoughts?
 
 
 --
 Juan Jose Comellas
 ([EMAIL PROTECTED])
 
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RE: [Asterisk-Users] How to use * and # as part of number in dial command

2005-08-28 Thread Michel Koenen

Hi Damon and others,

Your example is still doing what I tried already, so eventually the
dial command ends like:
Dial(zap/4/*21*)
or 
Dial(zap/4/*31*)
I prefer to use  Dial(zap/4/*21*thenumber)
or Dial(zap/4/*31*thenumber)

But whatever I try, the error message as in my first post shows up and
the line hangs up before the connection is made. So I assume the so
called 'keypad protocol' which is used here in the Netherlands and
possible in other countries is messing it up. I still hope there is
somebody out there who can help out.

Btw: my problem is not about how to use the Asterisk extension
(because that can be any extension where I want to put the dial
command after).

Best regards,
Michel Koenen

Damon Estep wrote:

* # are valid in a dialplan

you would start your exten = with the vertical service code *21*
then play prompt, collect digits, play prompt, dial
${exten}$(var_for_collected_digits}

BUT, unless I have missed something, You can just send *21* to the PSTN
and then follow their prompts! As long as DTMF is configured correctly
it should work. If they don't prompt you still dial the same way,
activate the vertical service code with *21*

Exten *21*,1,Dial(ZAP/CHAN/{$EXTEN})

That's all! Then just enter the rest of the digits, allowing the PSTN
switch to collect the DTMF and activate the code.

You may want to use something like this to specify which ZAP channel
your forward (or which line).

Add a line number first like this

Exten 1*21*,1,Dial(ZAP/CH1/*21*)

Notice we don't send the 1, but we do use it to pick the zap channel.
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[Asterisk-Users] All extensions now cannot loggin!!!!

2005-08-28 Thread Brian McCarey



Hi all,
I was gradually getting to grips with Asterisk at Home and hadgot things workingwith a voipuser account and three extensions (2 X-lites and a Sipura 2100) then suddenly all the extensions went down. Non will login...!
Have not got a clue why. Any ideas?
Also, as a side issue, I set up the voipuser account by setting up a blank trunk in AMP with the name 'voipuser'. I then set an outgoing route called 'voipuser' linking to the trunk. I then inserted the voipuser blurb in 'sip_additional.conf' and put a bit in 'extensions.conf' whichgot me up and running.The question is, If I follow the usual help stuff they tell me to insert the blurb in the sip.conf and extensions.conf. When I do this I cannot connect as ther is nothing in the AMP panel for trunks and extensions. The way i've done it with the blank trunk and out going route sets up the bits in the AMP panel. Why does the many help stuff never mention about doing anything with the trunk and outgoing route and instead only tells you to edit sip and extension config files Am I missing something??

regards
Brian

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Re: [Asterisk-Users] Good Deal on A Good Asterisk Box?

2005-08-28 Thread trixter http://www.0xdecafbad.com

On Sun, 2005-08-28 at 21:05 -0400, asterisk wrote:
 I am assuming two, couldn't a USB NIC be used?  Obviously not gigabit
 but can anyone see any problems with that setup?
  
USB throughput is less than max bandwidth which is what is advertised.
Add a hub and it gets even worse.  There is a substantial framing
overhead for usb. 

USB 1.1 has a raw transfer rate of 12Mbps
USB 2.0 has a raw transfer speed of 480Mbps

I believe the polling of USB devices is slightly more processor
intensive than of a pci card, but could be wrong (and then it may just
be the drivers that make it appear that way).

In theory 100Mbps wont have a problem on a USB 2.0 host, plenty of
bandwidth to spare, and depending on application it may be acceptable.
I however would not use such a device in a busy data center/colo for
fear that someone might unplug it (accidentally or intentionally) since
usb doesnt really lock in place.  


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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RE: [Asterisk-Users] How to use * and # as part of number in dialcommand

 Hi Damon and others,
 
 Your example is still doing what I tried already, so eventually the
 dial command ends like:
 Dial(zap/4/*21*)
 or
 Dial(zap/4/*31*)
 I prefer to use  Dial(zap/4/*21*thenumber)
 or Dial(zap/4/*31*thenumber)
 
 But whatever I try, the error message as in my first post shows up and
 the line hangs up before the connection is made. So I assume the so
 called 'keypad protocol' which is used here in the Netherlands and
 possible in other countries is messing it up. I still hope there is
 somebody out there who can help out.
 
I did not see an actual error message in your first post, what is the
error message?
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[Asterisk-Users] SER + ASTERISK voicemail

2005-08-28 Thread harry gaillac

Hello,

I try set Ua---SERAsterisk (voicemail/ARA)
|
   Ua
ser stable
asterisk cvs head 

I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to asterisk
voicemail in failure route.

How may I configure extensions.conf and ser.cfg ?
I have been trying without success!

Regards
Harry






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Re: [Asterisk-Users] Fedora Core 4 x86_64

2005-08-28 Thread Dennis Gilmore

Once upon a time Friday 26 August 2005 12:30 pm, Brian C. Fertig wrote:
 Take it from someone who owns 25 of them.  Stay away from FC anything.

 Use CentOS 4 its better more stable and has true multi-treading as FC
 doesn't thread anything..

What do you mean by FC doesnt thread anything?i have threaded applications 
usint NPTL  with over 1000 threads.  I really dont see how you can say FC is 
not stable  i have servers running FC  that have been up over 150 days.  the 
main issue i have has is the out of memory killer   going a little wild.  but 
i have had the same issue on some of my centos boxes.

I think they are both mostly as stable as the other.  tough CentOS base of 
RHEL  gives you a promise that the OS will be supported for a longer period 
of time.  This should mean  that you can run a CentOS based server for at 
least 5  years  without upgrading the server.

-- 
Dennis Gilmore  RHCE  
dennis AT ausil DOT us http://www.ausil.us


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Re: [Asterisk-Users] SER + ASTERISK voicemail

2005-08-28 Thread Steve Blair



You'll want some rules in your sip.conf to handle the connection from 
SER. A

starting point might be:

  [ser ip addr:ser port ?= 5060]
  type=peer
  context=my sip context name
  tos=lowdelay; tos delay
  allow=ulaw ; dtmfmode=inband only works with ulaw 
or alaw!

  dtmfmode=inband; Choices are inband, rfc2833, or info

You'll then want some rules in extensions.conf to accept the call and 
redirect it

to mailboxes defined in your voicemail.conf or in MySQL. Something like:

  [general]
  context=my sip context name
  switch = Realtime/my sip context name@extensions
  static=yes

 [my sip context name]

 exten = _uX,1,VoiceMail(${EXTEN}@my sip context name)
 exten = _X,1,VoiceMail(${EXTEN}@my sip context name)
 exten = _bX,1,VoiceMail(${EXTEN}@my sip context name))
 exten = #,2,Hangup ; Hang them up.

Steve

harry gaillac wrote:


Hello,

I try set Ua---SERAsterisk (voicemail/ARA)
   |
  Ua
ser stable
asterisk cvs head 


I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to asterisk
voicemail in failure route.

How may I configure extensions.conf and ser.cfg ?
I have been trying without success!

Regards
Harry






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Re: [Asterisk-Users] 1.2.0 Beta1


Anton Krall wrote:



Also, maybe it was me but I upgraded on a test server by doing the make and
make install over my cvs old one and when on the CLI I do a show version and
I get this:


CVS what? v1-0? HEAD?



server2*CLI show version
Asterisk  built by [EMAIL PROTECTED] on a i686 running Linux on
2005-08-27 22:54:47 UTC
server2*CLI


That is a bug in the tarballs for 1.2.0-beta1, I'll fix it tonight.


Also checked the show applications and saw no new ones there.. Did I
messed up upgrading over the old one?


Depends on what you upgraded from...
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Re: [Asterisk-Users] Detect Dialtone

2005-08-28 Thread Goran Dj.

Dialtone detection should be an option in .conf for zap channel, i agree
with that.

 Are you trying to play with the case where you have an analog phone
 bridged on your fxo line, and detect the lack of dialtone when
 someone is using that analog phone?

Belive or not, but at some places on the world are still in use some old
(non-digital) ATC-es which do now provide dial-tone instantly. For
example, when ATC ARF-102 is very congested with outgoing calls, you
must wait some (unknown) time to get dialtone (10sec, 1min, 5min...)


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Re: [Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd

cmisip wrote:
 I want to be able to send a dtmf key to asterisk and have mplayer
 forward or rewind.

pabx*CLI show application ControlPlayback
pabx*CLI
  -= Info about application 'ControlPlayback' =-

[Synopsis]
Play a file with fast forward and rewind

[Description]
ControlPlayback(filename[|skipms[|ffchar[|rewchar[|stopchar[|pausechar[|restartchar]]):
  Plays  back  a  given  filename (do not put extension). Options may also
  be included following a pipe symbol.  You can use * and # to rewind and
  fast forward the playback specified. If 'stopchar' is added the file will
  terminate playback when 'stopchar' is pressed. If 'restartchar' is added,
the file
  will restart when 'restartchar' is pressed. Returns -1 if the channel
  was hung up. if the file does not exist jumps to n+101 if it present.

  Example:  exten = 1234,1,ControlPlayback(file|4000|*|#|1|0|5)


-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

2005-08-28 Thread Soner Tari

I'd suggest turning off echotraining on the FXS altogether, and perhaps 
even
killing the echocanceller on FXS entirely.  (you won't be getting 
significant
echo from the FXS, and the FXO should be handling it anyway) -- 
echocancelwhenbridged might be an interesting thing to play with as well.


e.g. (assuming port 1-3 are FXO and port 4-7 are FXS)

echocancel=64
echocancelwhenbridged=yes
echotraining=800
channel = 1-3

echocancelwhenbridged=no
channel = 4-7


Andrew,

I am sure you know that in zapata.conf parameter settings are in effect 
until specifically overridden later on in the file. In the first paragraph 
you suggest that I turn off both echotraining and echocanceler on FXS 
channels, so may I correct your example, that is, do you mean something like 
the following?:


echocancel=64
echocancelwhenbridged=yes
echotraining=800
channel = 1-3

echocancel=no
echocancelwhenbridged=no
echotraining=no
channel = 4-7

Please correct me if I'm wrong, in your example echocanceler would still run 
on connections other than TDM (such as FXS-SIP). Did you knowingly mean it? 
With my additions above, FXS channels would never use echocanceler. Right?


Thank you guys for all the help and comments. Rich's last comments were 
quite enlighthening, as always. I never knew echocanceler could be used on 
FXS channels. Sorry for my ignorance (but nowhere in docs or wiki could I 
see this information, I should have thought about it, my bad).


I'll try and post the results.
Soner

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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

2005-08-28 Thread Steve Edwards


Since this is a lively topic, I'll ask here...

How can I measure the interval between the original and the echo?

On Mon, 29 Aug 2005, Soner Tari wrote:

I'd suggest turning off echotraining on the FXS altogether, and perhaps 
even
killing the echocanceller on FXS entirely.  (you won't be getting 
significant
echo from the FXS, and the FXO should be handling it anyway) -- 
echocancelwhenbridged might be an interesting thing to play with as well.


e.g. (assuming port 1-3 are FXO and port 4-7 are FXS)

echocancel=64
echocancelwhenbridged=yes
echotraining=800
channel = 1-3

echocancelwhenbridged=no
channel = 4-7


Andrew,

I am sure you know that in zapata.conf parameter settings are in effect until 
specifically overridden later on in the file. In the first paragraph you 
suggest that I turn off both echotraining and echocanceler on FXS channels, 
so may I correct your example, that is, do you mean something like the 
following?:


echocancel=64
echocancelwhenbridged=yes
echotraining=800
channel = 1-3

echocancel=no
echocancelwhenbridged=no
echotraining=no
channel = 4-7

Please correct me if I'm wrong, in your example echocanceler would still run 
on connections other than TDM (such as FXS-SIP). Did you knowingly mean it? 
With my additions above, FXS channels would never use echocanceler. Right?


Thank you guys for all the help and comments. Rich's last comments were quite 
enlighthening, as always. I never knew echocanceler could be used on FXS 
channels. Sorry for my ignorance (but nowhere in docs or wiki could I see 
this information, I should have thought about it, my bad).


I'll try and post the results.
Soner

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Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline   [EMAIL PROTECTED]Fax: +1-760-731-3000
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

2005-08-28 Thread Andrew Kohlsmith

On Sunday 28 August 2005 19:55, Soner Tari wrote:
Andrew sez:
  echocancel=64
  echocancelwhenbridged=yes
  echotraining=800
  channel = 1-3
 
  echocancelwhenbridged=no
  channel = 4-7

 I am sure you know that in zapata.conf parameter settings are in effect
 until specifically overridden later on in the file. In the first paragraph
 you suggest that I turn off both echotraining and echocanceler on FXS
 channels, so may I correct your example, that is, do you mean something
 like the following?:

 echocancel=64
 echocancelwhenbridged=yes
 echotraining=800
 channel = 1-3

 echocancel=no
 echocancelwhenbridged=no
 echotraining=no
 channel = 4-7

 Please correct me if I'm wrong, in your example echocanceler would still
 run on connections other than TDM (such as FXS-SIP). Did you knowingly
 mean it? With my additions above, FXS channels would never use
 echocanceler. Right?

Correct.  That's precisely how I meant to say it.  You may still want the echo 
canceller for Zap--SIP and so on since the FXS port is (potentially) the 
only hybrid in the circuit.

-A.
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Re: [Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd

2005-08-28 Thread trixter http://www.0xdecafbad.com

On Mon, 2005-08-29 at 11:37 +1200, Matt Riddell wrote:
 cmisip wrote:
  I want to be able to send a dtmf key to asterisk and have mplayer
  forward or rewind.
 
 pabx*CLI show application ControlPlayback

mplayer has advantages of more codecs as well, so you arent as limited.
In addition it will play tv (with tuner card), dvds, etc.  So you can
really pick what you want as your audio source.

It would seem to me to be not that difficult with an agi to use mplayer,
although I havent tried.

controlplayback seems to fit if all you want is mp3s however ...


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] OT: Are you using a Lucent?

2005-08-28 Thread Andrew Thrift


Hi Mathew,

We are interested in doing this too, is it possible you can share the 
information with us?


We are looking at using a TNT MAX to terminate 8 E1's from the Telco, 
but we need a way of receiving the SS7 signalling and passing it to the 
TNT's via IPDC or whatever.


Regards,



Andy

Matthew Boehm wrote:

Is anyone out there using Lucent brand equipment to handle an 
incomming DS3, converting all 672 calls to SIP (as G729) and sending 
those to Asterisk/SER over ethernet?


If you are and are willing to speak to my boss about your experiences 
(over the phone) with it, please contact me off list.


We have a possible contract with a local CLEC to handle their long 
distance, and they want to send to us using DS3 and SS7.


I'm trying to convince my boss to use a $9K Lucent, but he wants to 
spend much more by breaking out the DS3 into DS1's and stack up 6 
asterisk boxes with 1 4-port card in each.


Again, if you are using Lucent and are willing to speak to my boss 
about your experiences, please contact me off list so I can setup a call.


Thanks,
Matthew

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[Asterisk-Users] hfc-pci/zaphfc: Asterisk hangs with signalling bri_net_ptmp but not with bri_net

2005-08-28 Thread Ralph Aichinger


Please forgive me, if I misunderstand the problem completely.

Following instructions in several german blogs, I want to configure
Asterisk with a hfc-pci card, an old NTBA and an ISDN phone
as a SIP device.

It seems that I have to set signalling in zapata.conf to bri_net_ptmp.
When I do this, Asterisk will hang if started with -c, the last
lines of output being:

[res_features.so] = (Call Parking Resource)
 == Parsing '/etc/asterisk/features.conf': Found
   -- Registered extension context 'parkedcalls'
   -- Added extension '700' priority 1 to parkedcalls
 == Registered application 'ParkedCall'
 == Registered application 'Park'
 == Manager registered action ParkedCalls
 == Registered application 'HoldedCall'
 == Registered application 'AutoanswerLogin'
 == Registered application 'Autoanswer'
[chan_zap.so] = (Zapata Telephony w/PRI)
 == Parsing '/etc/asterisk/zapata.conf': Found
   -- Registered channel 1, PRI Signalling signalling
   -- Registered channel 2, PRI Signalling signalling
   -- Automatically generated pseudo channel

Here it hangs.

If I change signalling back to bri_net, the console
will appear.

If I try to connect to the hung Asterisk process
from another terminal with asterisk -r, everything
will look fine, except it seems that zap does not
work, e.g. the command help does not give me
the zap commands. show modules shows chan_zap
however:

monk:/etc/asterisk# asterisk -r
   -- Remote UNIX connection
Asterisk , Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk  currently running on monk (pid = 10973)
Verbosity is at least 5
monk*CLI help
   debug channel  Enable debugging on a channel
help  Display help list, or specific help on a command
load  Load a dynamic module by name
   logger reload  Reopens the log files
   logger rotate  Rotates and reopens the log files
no debug channel  Disable debugging on a channel
  reload  Reload configuration
   set debug  Set level of debug chattiness
 set verbose  Set level of verboseness
   show applications  Shows registered applications
show application  Describe a specific application
 show autoanswer  Lists autoanswer channels
   show channels  Display information on channels
show channel  Display information on a specific channel
 show config handles  Show Config Handles
   show dialplan  Show dialplan
   show file formats  Displays file formats
  show image formats  Displays image formats
show manager command  Show manager command
   show manager commands  Show manager commands
  show manager connected  Show connected manager users
show modules  List modules and info
show parkedcalls  Lists parked calls
   show switches  Show alternative switches
 show uptime  Show uptime information
show version  Display version info
 soft hangup  Request a hangup on a given channel
  unload  Unload a dynamic module by name
monk*CLI show modules
ModuleDescription  Use Count
chan_modem.so Generic Voice Modem Driver   0
chan_modem_aopen.so   A/Open (Rockwell Chipset) ITU-2 VoiceMod 0
res_musiconhold.soMusic On Hold Resource   1
res_features.so   Call Parking Resource1
chan_zap.so   Zapata Telephony w/PRI   0
monk*CLI

My zapata.conf:
;
; Zapata telephony interface
;
; Configuration file

[channels]
;
; Default language
;
;language=en
;
; Default context
;
;
switchtype = euroisdn
; p2mp TE mode
signalling = bri_net_ptmp
;signalling = bri_net

; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
;signalling = bri_net

pridialplan = local
;prilocaldialplan = local
;nationalprefix = 0
;internationalprefix = 00


echocancel=yes
echotraining = 100
echocancelwhenbridged=yes

immediate=no
overlapdial=yes
group = 1
context=default
channel = 1-2

My zaptel.conf:

# hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=nl
defaultzone=nl

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

dmesg snippet:
Zapata Telephony Interface Registered on major 196
PCI: Enabling device :00:0d.0 ( - 0003)
ACPI: PCI Interrupt :00:0d.0[A] - Link [LNKC] - GSI 10 (level, 
low) - IRQ 10
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd0864f00 fifo 
0xca778000(0xa778000) IRQ 10 HZ 1000

zaphfc: Card 0 configured for NT mode
zaphfc: 1 hfc-pci card(s) in this box.
Registered tone zone 3 (Netherlands)


Any Ideas?

TIA
/ralph
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Re: [Asterisk-Users] OT: Are you using a Lucent?

2005-08-28 Thread Andrew Thrift

We have the ability to do this on a large scale, but want to do it on a 
smaller scale for 1 to maybe a maximum of 5 TNT's.



Andrew Thrift wrote:


Hi Mathew,

We are interested in doing this too, is it possible you can share the 
information with us?


We are looking at using a TNT MAX to terminate 8 E1's from the Telco, 
but we need a way of receiving the SS7 signalling and passing it to 
the TNT's via IPDC or whatever.


Regards,



Andy

Matthew Boehm wrote:

Is anyone out there using Lucent brand equipment to handle an 
incomming DS3, converting all 672 calls to SIP (as G729) and sending 
those to Asterisk/SER over ethernet?


If you are and are willing to speak to my boss about your experiences 
(over the phone) with it, please contact me off list.


We have a possible contract with a local CLEC to handle their long 
distance, and they want to send to us using DS3 and SS7.


I'm trying to convince my boss to use a $9K Lucent, but he wants to 
spend much more by breaking out the DS3 into DS1's and stack up 6 
asterisk boxes with 1 4-port card in each.


Again, if you are using Lucent and are willing to speak to my boss 
about your experiences, please contact me off list so I can setup a 
call.


Thanks,
Matthew

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Re: [Asterisk-Users] 24 line softphone

2005-08-28 Thread Matt Florell

I wrote a php web page to do just this with astguiclient/VICIDIAL
across several servers at once. If you use or want to install
astguiclient let me know and I'll tell you more about how the code
works.

http://astguiclient.sf.net/

MATT---

On 8/28/05, Ben Brown [EMAIL PROTECTED] wrote:
  I have no problem joining the conferences and monitoring. What I need is a
 nice, simple, preferably GUI method to switch between multiple active
 connections. I have a method I like using a 3 line softphone, which works
 for 3 conferences, but I need one line for each connection to use my
 method, thus I am seeking a 24 line softphone. 
  
  Thanks for the thoughts
  
  BEN
  
 
  Steve Edwards wrote: 
 On Sun, 28 Aug 2005, Kevin P. Fleming wrote: 
  
  
 Ben Brown wrote: 
  
  
 I suppose if there was just a way to monitor the 24 conferences on request,
 then the participation could be accomplished using a regular SIP client. 
  
  In CVS HEAD (and soon Asterisk 1.2), app_chanspy will do exactly that. 
  
  Chanspy will not let you join a conference, only spy. 
  
  I do something similar with this snippet. 
  
  [supervisor] 
  ; monitor an agent 
  exten = s,1,   
 background(please-enter-a-conference) 
  exten = _xx#,1, meetme(${EXTEN:0:6},ampqs) 
  exten = _xx#,n, goto(${CONTEXT},s,1) 
  
  You enter a conference number and #, to join with mute on. You can 
  *1 to unmute if you want to speak. Press # to exit the conference 
  and you are prompted to enter another conference number. 
  
  I've hacked on app_meetme a bit to control who hears enter and leave 
  sounds and the application exit code as well as let the supervisor 
  kick all users or admins. 
  
  Thanks in advance, 
 
  Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST 
  Newline   [EMAIL PROTECTED]Fax: +1-760-731-3000 
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Re: [Asterisk-Users] 1.2.0 Beta1

2005-08-28 Thread David Liu


Hi Kevin,

Just courious how to fix the bug for not a blank space in Asterisk 
version number?  I think I have seen that in the past but downloading a 
new copy from CVS fixes it.  No harm but it tickles.


David

Kevin P. Fleming wrote:


Anton Krall wrote:


Also, maybe it was me but I upgraded on a test server by doing the 
make and
make install over my cvs old one and when on the CLI I do a show 
version and

I get this:



CVS what? v1-0? HEAD?



server2*CLI show version
Asterisk  built by [EMAIL PROTECTED] on a i686 running 
Linux on

2005-08-27 22:54:47 UTC
server2*CLI



That is a bug in the tarballs for 1.2.0-beta1, I'll fix it tonight.


Also checked the show applications and saw no new ones there.. Did I
messed up upgrading over the old one?



Depends on what you upgraded from...
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RE: [Asterisk-Users] 1.2.0 Beta1

2005-08-28 Thread Anton Krall

I upgraded from cvs head 1.0.x which in my case was cvs head about 2 months
ago.

Do you recommend doing a clean install vs. installing on top?

What new dialplan features should I look for and new apps on 1.2.0 beta1?  

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kevin P. Fleming
|Sent: Domingo, 28 de Agosto de 2005 05:58 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] 1.2.0 Beta1
|
|Anton Krall wrote:
|
|
| Also, maybe it was me but I upgraded on a test server by doing the 
| make and make install over my cvs old one and when on the CLI I do a 
| show version and I get this:
|
|CVS what? v1-0? HEAD?
|
| 
| server2*CLI show version
| Asterisk  built by [EMAIL PROTECTED] on a i686 running 
| Linux on
| 2005-08-27 22:54:47 UTC
| server2*CLI
|
|That is a bug in the tarballs for 1.2.0-beta1, I'll fix it tonight.
|
| Also checked the show applications and saw no new ones 
|there.. Did I 
| messed up upgrading over the old one?
|
|Depends on what you upgraded from...
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|

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[Asterisk-Users] Asterisk 1.2.0-beta1 tarball re-released


Due to a packaging error, the tarball released on Friday night did not
have a version number embedded in it, which results in various strange
build errors and other odd behavior.

The tarball on the FTP servers has been updated to correct this
situation. Sorry for the inconvenience :-)

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[Asterisk-Users] error messages

2005-08-28 Thread Chris Wilson


Hey,

does anyone know why i'd be receiving:

Aug 28 19:40:04 DEBUG[1875]: # Testing 66.27.233.241 with 10.0.10.0
Aug 28 19:40:04 DEBUG[1875]: Target address 66.27.233.241 is not local, 
substituting externip


I get tons of them, usually when the phone is registering/calling/receiving 
calls.


Thanks!

Chris 


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[Asterisk-Users] Japanese ISDN BRI card for asterisk (INS64) where to start?

2005-08-28 Thread Mick Hastings

Hi All,

I currently run asterisk in our office (in Japan) and use a cisco PRI 
gateway for connection to the PSTN. I would like to setup some more systems 
for our smaller offices (in Japan) that would use BRI and preferably using a 
PCI card in the asterisk box and not a seperate Cisco gateway (expensive). 
HOWEVER, Japan has this INS64 protocol for their BRI lines and im not sure 
what cards are available that are compatible with asterisk and Japanese BRI 
(INS64). I know that it is supported by Cisco (like they support Japanese T1 
PRI (INS1500)) but it just adds to the cost and is another piece of 
hardware.

I tried searching the archives and only found a few references to INS64 and 
it didnt sound too promising. I then searched the net and found this 
Intel/Dialogic board:

BRI/80-PCI BRI/PCI Series High-Density ISDN Basic Rate Interface Boards
(for details see: http://www.intel.com/network/csp/products/7007web.htm)

It seems to support INS64 but appears to only have windows drivers. Has 
anybody used this cards with asterisk? is it possible? or even likely that 
it would be supported by any of the linux ISDN drivers?

I also noticed some other mentions of 'ISDN protocol converters' What are 
these specifically? (im guessing they convert between US BRI standards and 
INS64), how much are they? where do I get one?

Has anybody out there got an asterisk system running with INS64 connections 
to their box? If so could you please let me know how you are doing it, else 
can anybody offer any information as to where I should start to look for 
more informaion this topic?

I really appreciate the help.

cheers,
Mick Hastings







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Re: [Asterisk-Users] 1.2.0 Beta1


Anton Krall wrote:

I upgraded from cvs head 1.0.x which in my case was cvs head about 2 months
ago.


There is no such thing as cvs head 1.0.x. You could mean 'CVS v1-0 
(whatever was current in the 1.0.x branch at the time) or 'CVS HEAD' 
(the current development branch at the time).



Do you recommend doing a clean install vs. installing on top?


Unless you were running a recent CVS HEAD already, yes, a clean install 
is a good idea.

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Re: [Asterisk-Users] Mplayer as replacement to mgp123 in MP3Player cmd

trixter http://www.0xdecafbad.com wrote:
 controlplayback seems to fit if all you want is mp3s however ...

Although it works with all supported formats.

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards

Adam Robins wrote:
 We are in the process of an Asterisk call center deployment using IAX2
 G711 ulaw softphones.   Outbound sound quality is terrible.  

Check if the network card is in half duplex mode.

-- 
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Re: [Asterisk-Users] Need quote for Asterisk and billing remote install