[asterisk-users] How to read DTFMs from MEETME_AGI_BACKGROUND without blocking?

[Cosmin Prund]

Hello everyone.

 

What I'm doing: 

 

I've made a replacement for app_queue that uses MeetMe to connect the
calling party with the agents. When the call comes in it gets put into a
MeetMe room with a nice AGI_BACKGROUND so the calling party can listen
to music and announcements until an agent becomes available. So far
everything works fine. Now I want to give the calling party an one-digit
menu (press 0 for an operator, 1 for accounting, etc) so he can
automatically switch queues. I can do that using the read application
but the read application is blocking: it will not return until the
digit is pressed OR the timeout expires. Unfortunately this is not
optimal because the background AGI needs to stop when an agent becomes
available. So I'm looking for a way to make digits available in the
background AGI *without* blocking the background AGI!

 

I already tried using features.conf mapped digits. They don't seem to
work with MeetMe. I've made a simple one-digit entry that calls
Verbose to output something to the CLI. It works fine in a normal call
but it doesn't work when the call is in the MeetMe room.

 

I looked for Asterisk manager events that might get fired when DTFMs get
pressed. Unfortunately I found none. Do such events get fired? If so -
how do I enable that?

 

What I'm asking: Is there a way to receive DTFM digits asynchronously?
Or to get features.conf appmap's to work in a MeetMe room? Or to get
Asterisk to fire manager events when DTFM's are pressed?

 

Thanks,

Cosmin Prund

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Re: [asterisk-users] Tracking Call Time While in Dial()

[Cosmin Prund]

Call an AGI right before the start of the Dial command to record the start 
time and ether use an manager application (makes use of manager API) or call an 
DeadAGI once the call has ended (from the h extension). This requires a bit 
of programming - but then again some programming is required anyway to display 
the actual talk time somewhere. It might also be that I'm an programmer and I 
attempt to solve all problems writing programs, so maybe someone else has a 
better idea!

 

--

Cosmin Prund

 

De la: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] În numele Douglas Garstang
Trimis: Thursday, July 10, 2008 7:49 PM
Către: asterisk-users@lists.digium.com
Subiect: [asterisk-users] Tracking Call Time While in Dial()

 

So, I've been asked if this is possible.

Someone wants to actively monitor the duration of a call, while the call is 
still in progress. Obviously, in Asterisk, once the Dial() application starts, 
you lose dial plan control until after the call has ended, successful or 
otherwise.

Anyone know if that kind of thing is possible?

Doug.

 

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Re: [asterisk-users] Zap Bridged Channels

[Cosmin Prund]

I've had a similar problem with a A101DX + A202DX. I was trying to bridge from 
my A101 to my A202 to get faxes over my E1 line. I've done an number of things, 
I'm not exactly sure which one helped, but it now works very nice for fax:

 

(1) Using zap show channel N on the CLI I noticed that echo canceling was on 
even those I was bridging two Zap devices. I disabled the HWEC on my A202 card 
and it's now ok (no echo cancel on the A202 card). This was an option for me 
because I'm only doing fax on the analog card. I don't think this had a lot to 
do with the final fix of the problem.

(2) I emailed Sangoma and they told me there's a newer version of the drivers 
that tweek the echo cancel algorithm to make it better suited for fax. The 
driver was beta at the time so I didn't try it. You might want to contact 
Sangoma yourself!

(3) I fixed my zttest timing! When I tested I had really bad timing (94,00 
worst and  99,00 average). The docs and the wiki say that's bad timing but I 
had absolutely no problems with voice quality. None! And I've only done one 
thing to fix my timing: /etc/init.d/irqbalance restart. I have no idea why 
that makes a difference but it does and I've now got 99,95 average timing from 
zttest with the worst being over 99,00. You might want to try this yourself 
since you also seem to have the X (pci express) version of the card.

 

--

Cosmin Prund

 

 

De la: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] În numele Jeremy Mann
Trimis: Wednesday, July 09, 2008 10:28 PM
Către: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subiect: [asterisk-users] Zap Bridged Channels

 

I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS for 
modem connectivity.

 

I have Zap/8 as a Fax Machine

 

Zap/5 is my outside line.  When a call rings in on Zap/5 it immediately calls 
Zap/8 and bridges the channels.  I see it doing a native bridge on the two.  I 
have echo cancel off on native bridge, but I can never get fax connectivity, it 
just tries to negotiate forever then eventually hangs up.

 

Anything special to getting this to work?  

 

Below is an example of CLI output when the Fax Machine tries to call out, it 
does the same thing, never get the two machines to complete the call and send 
the fax.  I've also included the CLI output of channel 5's properties, it does 
show the EC as off.  I noticed it says Fax Handled: no, is there something I 
need to enable in Zapata.conf or zaptel.conf?

 

Would txgain/rxgain be the issue?

 

CLI Output 

-- Starting simple switch on 'Zap/8-1'

-- Executing [EMAIL PROTECTED]:1] Answer(Zap/8-1, ) in new stack

-- Executing [EMAIL PROTECTED]:2] Dial(Zap/8-1, Zap/5) in new stack

-- Called 5

-- Zap/5-1 is ringing

-- Zap/5-1 is ringing

-- Zap/5-1 answered Zap/8-1

-- Native bridging Zap/8-1 and Zap/5-1

 

localhost*CLI zap show channel 5

Channel: 5CLI

File Descriptor: 27

Span: 2

Extension:

Dialing: no

Context: from-internal-fax

Caller ID:

Calling TON: 0

Caller ID name:

Destroy: 0

InAlarm: 0

Signalling Type: FXO Kewlstart

Radio: 0

Owner: Zap/5-1

Real: Zap/5-1

Callwait: None

Threeway: None

Confno: -1

Propagated Conference: -1

Real in conference: 0

DSP: yes

Relax DTMF: yes

Dialing/CallwaitCAS: 0/0

Default law: ulaw

Fax Handled: no

Pulse phone: no

Echo Cancellation: 128 taps unless TDM bridged, currently OFF

Master Channel: 8

Actual Confinfo: Num/8, Mode/0x0009

Actual Confmute: No

Hookstate (FXS only): Onhook

 

Zapata.conf -

 

[channels]

context=default

usecallerid=yes

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

canpark=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=no

relaxdtmf=yes

rxgain=2.0

txgain=2.0

group=1

callgroup=1

pickupgroup=1

immediate=no

context=from-internal-fax

group=1

signalling = fxo_ks

channel = 5

context=from-zaptel-fax

group=3

signalling = fxs_ks

channel = 8

 



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Re: [asterisk-users] asterisk ivr

[Cosmin Prund]

Your best option is to use queues. If for some raison you can't use queues 
you'll need to do some serious programming (agi, manager api) to get things 
working. You can probably do the basic stuff using dialplan logic and a few 
shell scripts, but you'll need to get a lot more involved when you'll need to 
deal with errors.

Easy sample: You can send the originating caller to a auto-created MeetMe room 
where he/she will listen to music. You can somehow start an ORIGINATE command 
and have your side of the call (the agent) go into the same MeetMe room. You 
can program the MeetMe room to automatically tear down when your agent hangs up 
(you make your agent an marked user and you make the MeetMe room for the 
originating caller to automatically exit when the marked user exists). You can 
train your agent to hang up when he starts hearing MOH (the calling user hang 
up). But after all this simple stuff you'll need to start dealing with errors: 
What happens if your agent doesn't answer the call placed by Asterisk? Your 
calling user would stay in the MeetMe room listening to music for ever!

If you do want to go the hard way (agi+manager api) it can be done, and I've 
done it. I'm using a Delphi application that handles both the agi stuff (using 
FastAGI) and the manager stuff. This allows me to interact with Asterisk in 
more ways then one way (I've got permanent access to the CLI so I can send 
originate commands, I can kick people from conference rooms to make sure they 
don't stay there forever). I've basically re-implemented the queue stuff in my 
own code, but I've done it because the basic queue stuff in Asterisk can't 
readily apply to my scenario: My Agents do other things beside talking on phone 
so I can't force them to always answer the phone and I wanted to give them a 
lot of information about the calling clients before they actually take the 
call! 

P.S: I'm not an Asterisk guru. I'm not in the business of selling Asterisk. My 
experience comes from implementing Asterisk for my own organization - so it's 
limited experience. 

--
Cosmin Prund

 -Mesaj original-
 De la: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] În numele Philipp Ott
 Trimis: Monday, July 07, 2008 3:10 PM
 Către: Asterisk Users Mailing List - Non-Commercial Discussion
 Subiect: [asterisk-users] asterisk ivr
 
 Hello!
 
 We would like to receive a SIP call and keep the caller waiting
 listening to some music other sound. A secondary intelligence decides
 whom to connect to and creates an outbound SIP call and when it is
 ringing there, or after the recipient answered the call, and maybe
 after
 listening to some small IVR joins the waiting caller, thus cancelling
 the music.
 
 Although the DIAL command offers many many options and we can put all
 the intelligence of whom to connect to whom there (or in scripts) we
 have the problem that the music always starts from the beginning when a
 new DIAL is started. This isnt an elegant solution. So the idea we got
 was to keep the caller in a meetme conference of 2 people. But how then
 can we force asterisk to dial out (most likely a secondary asterisk
 invocation with a rx command), make it go through some minimal
 context/dialplan upon answering, and eventually connect the called
 person to the meetme conference of the incoming call? Naturally, all
 this without any pin-codes or such.
 
 Did anybody have this problem already and maybe even found a solution
 for it?
 
 Thank you
 Regards
 Philipp
 
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[asterisk-users] Replace music-on-hold on MeetMe with ringing sound

[Cosmin Prund]

Hello. It's been a while since I last posted (probably because my * works 
just fine). I'm working on something to replace call queues in my own 
application-specific way and I'm using MeetMe rooms to bridge agents and 
clients and do other things.
 
When an agent needs to be bridged with a client I'll first put the agent in the 
MeetMe room and when I have confirmation that the agent is in the MeetMe room 
I'll send the client to the same room. My agent gets to hear music on hold 
while it's the only one in the conference room (it takes 1 or 2 seconds for the 
client to be put in the same room). Is it possible to make the agent here 
ringing (or replace the music on hold with a recording of ringing)? 
 
At the moment I'm telling agents when the music stops playing you're talking 
to the client but that just doesn't sound right and it's a bit fiddely because 
music on hold is music and music has pauses. One can imediatelly tell the 
ringing is done but they might need a few extra seconds to realise the music 
has stoped. On the other hand the client has no such problem since he/she hears 
ringing just before they get bridged to the MeetMe room.
 
Any ideas? Thanks!
 
--
Cosmin Prund
 
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[asterisk-users] Softphone that emulates Skype API ?

[Cosmin Prund]

There's a large number of gadgets one can buy that work with Skype
through the API. One of the things I'm interested right now is the
ability to properly use a mobile phone headset with a SIP/IAX softphone.

 

Is there an softphone that emulates the Skype API?

Are there legal implications in writing an softphone that emulates the
Skype API?

Should I just give up and buy a Siemens DECT phone that supports a
bluetooth headset?

 

--

Thanks,

Cosmin Prund 

 

 

 

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Re: [asterisk-users] About .call files when the congestionis on myside

[Cosmin Prund]

 Behalf Of Anselm Martin Hoffmeister wrote:
 Sent: 16 octombrie 2007 09:29
 Subject: Re: [asterisk-users] About .call files when the congestionis
 on myside
 
 
  *IF* an unanswered call stops the retry cycle then it's true, I can
 simply
  ask for lots of retries. I assumed an unanswered call would NOT stop
  the retry cycle so I was afraid to set a large value here. I'll have
  to test what happens if the called line doesn't pick up the phone.
 
 An unanswered call should just initiate another Wait, followed by a
 retry. Unanswered means as much as unsuccessful, for the purpose of a
 call file is to dial out and get whatever done.
 
 If you want unanswered calls to be successful (which does not make much
 sense to me, because the fax has not been delivered), you probably need
 scripts that do the management for you.
 

What I really want is to get a real chance for the fax to go throw, and I'm 
looking for some balance here. The way call files seem to work out of the box 
is absolutely perfect IF the lines are not very congested. Since in my case the 
congestion is on my side most faxes don't even get 1 real try (they all fail 
because my side is congested). For me a good solution would be one where the 
call is not counted if the congestion is on my side. A perfect solution would 
be one where the try starts and loops till a local line becomes available - 
so I can work with acceptable wait times between tries.

Failing all that, I had hope from a previous post: IF an un-picked remote 
ringing phone would stop the retry loop, I could use a short wait between tries 
and a large number of tries (so I'd try and try till the phone rings at least 
once). But that doesn't work, since an unanswered ringing phone doesn't stop 
the retry loop. I don't want to risk making an customer's phone ring for 2 
hours non-stop just because there's no one near the phone to pick it up or the 
fax went out of paper and refuses to auto-answer!

Now I'm left with 3 options:

(a) Hope for a solution/tip from the list.
(b) Some kind of management-api based solution.
(c) A code-hack.

Out of the three, the first option is probably best (I'm obviously no 
Asterisk-guru). (b) might work well as I might be able to actually loop till I 
send the fax, not till I get an answer; (c) might also work, but I sure hope I 
don't need to go there...

--
Cosmin Prund
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[asterisk-users] Best sotphne to se ith a BlueTooth Hadset, a PC and a USB dongle

[Cosmin Prund]

Hello everyone.

 

I recently boght a Nokia BH900 headset and USB bluetooth dongle and I'd
like to use them to make calls from a sofphone. I managed to this with
boxe XTen-Lite and the Zoiper - but they both see the device as a simple
sound card through the BlueSoleil drivers. While this is allmost usable,
the headphone seems to be kept in transmission mode all the time and I
get a constant hi in the headset when I'm not actually on a call.
Also the answer button on the headset doesn't work as it would work if
hooked up to a mobile phone.

 

Is there a softphone that can make proper use of a bluetooth headset,
free or comercial?

 

--

Thanks,

Cosmin Prund

 

 

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[asterisk-users] About .call files when the congestion is on my side

[Cosmin Prund]

Hello everyone.

 

I'm working on an application that needs to automatically send faxes. To
send the faxes I create .call files but the .call files mostly fail
because my lines are always congested within business hours! Is there
any trick I can use to give the end user a better chance at actually
receiving the faxes?

 

I already tried using the local channel for dialing (so I can put in
there a loop that waits for a line to be available) but this doesn't
work because I'm sending faxes using chan_capi's capicommand(sendfax) -
and that command requires an chan_capi channel, it doesn't like the
local channel. Besides, looping in the dialplan would probably
interfere with the Wait option in the .call file so that's a really
bad solution. 

 

--

Thanks for any suggestion,

Cosmin Prund

 

 

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Re: [asterisk-users] About .call files when the congestion is on myside

[Cosmin Prund]

 Behalf Of Anselm Martin Hoffmeister wrote:
 Subject: Re: [asterisk-users] About .call files when the congestion is
 on myside
 
 Am Montag, den 15.10.2007, 11:26 +0300 schrieb Cosmin Prund:
  Hello everyone.
 
 
 
  I’m working on an application that needs to automatically send faxes.
  To send the faxes I create .call files but the .call files mostly
 fail
  because my lines are always congested within business hours! Is there
  any trick I can use to give the end user a better chance at actually
  receiving the faxes?
 
 Are you aware of the MaxRetries, RetryTime and WaitTime in your
 call-files? You can set quite large numbers, e.g. a RetryTime of
 15 minutes and a MaxRetries of 32 would try for up to 8 hours.
 
 Note though that any answered call will stop the retry cycle.
 This is embarassing for Zap channels that cannot detect remote
 ringing / remote busy reliably. As you use ISDN this should not
 be a problem

*IF* an unanswered call stops the retry cycle then it's true, I can simply ask 
for lots of retries. I assumed an unanswered call would NOT stop the retry 
cycle so I was afraid to set a large value here. I'll have to test what happens 
if the called line doesn't pick up the phone. 

  I already tried using the local channel for dialing (so I can put in
  there a loop that waits for a line to be available) but this doesn’t
  work because I’m sending faxes using chan_capi’s capicommand(sendfax)
  – and that command requires an chan_capi channel, it doesn’t like the
  “local” channel. Besides, looping in the dialplan would probably
  interfere with the “Wait” option in the .call file so that’s a really
  bad solution.
 
 If you want to do this (looping) use MaxRetries = 0. I do not
 understand
 why having the remote side connecting to a local extension that does
 faxing would not work. Or is it that the CAPI FAX stuff will only work
 on unAnswer()ed channels?

It's the CAPI stuff not wanting to send over a non-CAPI channel. And it somehow 
makes sense, because the CAPI stuff uses the DSP's in my ISDN card, so it can't 
work unless it's on a CAPI channel. Also I expected the capi application to see 
through the Local channel and notice it really is an CAPI channel!


Thanks for your answeres!
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RE: [asterisk-users] NAT

[Cosmin Prund]

NAT is not that big of a problem, not anymore.
Do a NAT search on http://www.voip-info.org - it'll get you started (got me 
started at least)

--
Cosmin Prund

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Henry Cobb
 Sent: Tuesday, June 05, 2007 4:30 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] NAT
 
 On 6/5/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote:
  Hi All!!
 
  I have my asterisk working in my house (working with mandriva 2007
 and
  asterisk 1.4 svn). I´ve looking on the net but i couldn´t find the
 way of
  making work my asterisk in a real enviroment. Seems that the problem
 of NAT
  is a big problem. How can I sort out this, a mean crossing the NAT
 and
  having asterisk connected?
 
 If you want to receive calls and not just place them and you have a
 broadband connection with a dynamic IP then your server must register
 with the VoIP provider and I suggest using IAX with the proper UDP
 port assigned to your Atrisk server.
 
 -HJC
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RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware

[Cosmin Prund]

Keep in mynd, SIP requires a stable timing source. Don't know how Xen handles 
timing, but with vmware you can get all sorts of issues with timing: the clock 
goes faster or slower then normal on multi core systems and on systems with 
power stepping.

In my case i'm getting  those timing issues on two dual core amd machines and 
i'm not getting timing issues on three dual-core intel   machines.

--
Cosmin Prund


-Original Message-
From: Adam Robins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: 29.05.07 18:09
Subject: RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware

Thanks, but we do not use any zap hardware in these systems.  It is straight 
SIP.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François 
Delawarde
Sent: Tuesday, May 29, 2007 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware

Hi,

Be careful with believing too much that your zaptel hardware will work 
together with xen, you could have problems like the ones described in 
the thread linked below:

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg180825.html

Good luck,
François.



Adam Robins wrote:
 We are running Asterisk on native CentOS.  We then install VMWare on
 CentOS with Windows 2003 in the VMWare partition for AD services.  We
 have 50+ users in a call center environment with no issues.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
 Creasy
 Sent: Sunday, May 27, 2007 11:52 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware

 Why would you want to do this?

 If you wanted to run multiple systems together on an Asterisk server I 
 would run the Asterisk server on Dom0 and the other stuff on DomU
 systems.

 -Jonathan

 James Harper wrote:
   
 I did it back in the xen 2.x days with a BRI adapter (Traverse
 
 NetJet).
   
 It worked fine for the testing I was doing.

 I'm not sure of the status or performance of the PCI mapping through
 
 to
   
 DomU these days, but that should be the only extra step required.

 James

   
 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Roberto Pereyra
 Sent: Saturday, 26 May 2007 23:06
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Asterisk in Xen domu with tdm400 hardware

 Hi all !!!

 I would like to install asterisk in Xen domU using TDM400 hardware.

 Somebody know a howto or tutorial about that ?

 Thanks in advance

 roberto

 --
 Ing. Roberto Pereyra
 ContenidosOnline
 http://www.contenidosonline.com.ar
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RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware

[Cosmin Prund]

There was a discusion on the subject a few days ago, search  the archives . The 
quick answer is you don't, but don't take my word for it, I know nothing 
about xen and very little about asterisk!

--
Cosmin  Prund 

-Original Message-
From: Roberto Pereyra [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: 26.05.07 16:09
Subject: [asterisk-users] Asterisk in Xen domu with tdm400 hardware

Hi all !!!

I would like to install asterisk in Xen domU using TDM400 hardware.

Somebody know a howto or tutorial about that ?

Thanks in advance

roberto

-- 
Ing. Roberto Pereyra
ContenidosOnline
http://www.contenidosonline.com.ar
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RE: [asterisk-users] transfer call sip to zap

[Cosmin Prund]

It just works. Simply transfer your call to the desired extension and
let Asterisk take care of the details.

 

--

Cosmin Prund

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of DiegoF
Sent: Friday, May 25, 2007 12:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] transfer call sip to zap

 

how to transfer a call from sip channel to zap channel

thanks

-- 
//  DiegoF  //

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RE: [asterisk-users] Working softphone for poket PC

[Cosmin Prund]

This is my SJphone story, this is why I removed it:

I installed SJphone without too much trouble, I found a voip-info
article on configuring it and tried configuring it. Apparently I failed
to configure it properly since it did not attempt to register to my
asterisk server (in fact, selecting the asterisk profile would do
nothing, it would simply jump right back to the pc-to-pc sip profile).
So I tried fixing the configuration - failed to that because the
Options menu option failed to work! Every single other option would
work, but NOT that one!

So I uninstalled it :-)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Post
Sent: Tuesday, May 22, 2007 11:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Working softphone for poket PC

Cosmin Prund wrote:
 Googling arround I found a number of pocket pc softphones. Of those I
 was only able to install SJ-something (removed it).
 

SJphone, and why did you remove it?

 Is there one (pocket pc softphone) that works?
 

SJphone ;-) At least I've made some successful calls using sjphone

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-- 

Remco Post

I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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RE: [asterisk-users] Working softphone for poket PC

[Cosmin Prund]

Do you remember anything else about the Microsoft thingy from the
developer resources or whatever so I can google for it a bit? Anyway,
not working reliably is not going to stop me, since I really don't
expect it to work reliably! But being able to use my PDA to make an
_TEST_ call would be really cool.

Guess I agree with you, this is another men's toy, since it really has
no practical use to it.

--
Cosmin Prund 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: Tuesday, May 22, 2007 11:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Working softphone for poket PC

Am Dienstag, den 22.05.2007, 21:49 +0300 schrieb Cosmin Prund:
 Googling arround I found a number of pocket pc softphones. Of those I
was only able to install SJ-something (removed it).
 
 Is there one (pocket pc softphone) that works?

When I searched for one, about half a year ago, there were two that
actually worked, but both had their flaws. One was SJphone, and that was
hard to get running. The other one was a Microsoft thingy, from their
developers ressources or whatever, that always used the loudspeaker
instead of the earphone piece...

Somehow they worked, but back then, I decided against and got a separate
WLAN phone from ebay. Not that that turned out to work more reliably,
mind, but at least some more men's toys ;-)

I would be glad to learn about a Wince softphone that actually worked
without choking on something like a phonenumber callerid starting +,
or just the random PDA crash that makes the reset button wear out.

Best,
Anselm

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RE: [asterisk-users] WiFi SIP phones

[Cosmin Prund]

I'm personally interested in one of those VoIP/DECT phones (where the
VoIP is handled by the base and the base-connection is wired) but I
wander if they are better than a standard DECT phone + an ATA (I've
already got two DECT phones pluged into ATA's around the office + 1
@home and I know for sure they work really well).

Specifically: Does the Siemens support rejecting an VoIP call? Does it
start ringing  showing Caller info immediately (as opposed to the
ATA+DECT combination, as that requires 1 ring to get caller info)

Thanks,
Cosmin Prund

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
mailinglist
Sent: Wednesday, May 23, 2007 2:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] WiFi SIP phones

I'd just like to say that I purchased a siemen S450IP recently and so
far so
good it's a nice handset and works better than previous wifi phones I've
used. This is most likely due to it being dect gap where the base
station
handles the voip side and not the phone thus avoiding issues with 802.11
wireless and phone packets.

Regards,
Dee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Bagnall
Sent: 23 May 2007 11:02
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] WiFi SIP phones

Greetings list,

What are people's experiences with WiFi SIP phones?

When I last looked into them about 18 months ago, they were incredibly
expensive, had very limited range and poor battery life. In the end, it
worked out much more cost effective to simply use ATAs + DECT cordless
phones where there was a requirement for portable devices.

I assume things must have moved on somewhat since then. What models are
currently out there people would recommend I look at?

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons

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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date:
22/05/2007
15:49
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date:
22/05/2007
15:49
 

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RE: [asterisk-users] WiFi SIP phones

[Cosmin Prund]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Remco Post
 Sent: Wednesday, May 23, 2007 10:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] WiFi SIP phones
 
 Tony Plack wrote:
  Are the DECT phones two channel or do they share a channel like most
  other portable phones?
 
 DECT is a digital standard, quite distantly comparable to GSM. There
 are
 multiple channels (I believe the standard allows for 12 channels, but
 the last time I actually worked on DECT is ages ago). A siemens S450IP
 can have up two 6 handsets with 2 'external' (SIP or POTS) phonecalls
 concurently. You cannot decline a phonecall, but you can ignore it.

I'm curious: is the impossibility of declining an call a DECT limitation
or is it that agest ago DECT phones were backed by POTS line and
declining a call would make no sense since the POTS doesn't support it? 
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[asterisk-users] Working softphone for poket PC

[Cosmin Prund]

Googling arround I found a number of pocket pc softphones. Of those I was only 
able to install SJ-something (removed it).

Is there one (pocket pc softphone) that works?

Thanks,
Cosmin Prund___
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RE: [asterisk-users] zaptel huge irq problem

[Cosmin Prund]

I'm no Asterisk expert (nor Linux expert) but I am using my * box for 
multiple things (transparent firewall, NAT box, samba server, poptop server) 
and for a considerable time I've been running a VmWare server with a Windows XP 
virtual machine up-and-running at all times! The Windows XP VM was running IIS, 
Apache, WarFTP and a Firebird database server - all of which got moderate use.

The hardware for my * box is what would be considered moderate-to-cheap: 
Sempron-something processor (not a big processor, don't remember the exact 
GHz), enough RAM (I've added 1 Gb of RAM when I've started using VmWare 
server), a nice motherboard (I remember I specifically looked for a motherboard 
with the minimum amount of on-board devices, of which I have disabled 
everything I don't need!). The extra hardware on my box includes 2 PCI NIC's 
(I'm also using the on-board NIC so I've got 3 working NIC's), an TDM400 card 
with 3 FXO and 1 FXS, and an Diva Eicon Server BRI card for my ISDN connection. 
I've got 3 HDD's into the box, of which 2 are old IDE drivers (parallel ATA) 
and the other one is SATA.

My VoIP experience has been good, my zaptel timing is pretty good and I can get 
faxes working on the FXS interface as well (coming in over the ISDN line).

The rationale behind placing the Windows XP virtual machine on the * has not 
been the lack of extra hardware but the desire to keep the number of always-on 
servers to a minimum. I've since moved the VM off the Asterisk server because 
I've installed an Windows SBS 2003 server on a considerably more powerful 
server.

So there it goes, proof that a small-office Asterisk box can do lots and lots 
of things!

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François 
Delawarde
Sent: Thursday, May 17, 2007 8:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] zaptel huge irq problem

Hi,
 Why are you so determined to use Asterisk in a VM? You're asking for
 trouble. Asterisk belongs on dedicated hardware.
   
I actually want to use Asterisk in a machine HOSTING a VM (that's what I 
implied with the Dom-0 thing I said earlier), sorry for the 
misunderstanding. I agree with you that given the state of advancement 
of just about any 'virtualizer', I would have to be totally stupid to 
try running Asterisk inside a VM. (I also wouldn't have asked here in 
the first place, as I would have been totally certain that problems came 
from the virtualizer itself)

If you feel concerned with my reasons for doing that anyway:

- No one told me that Asterisk belonged on dedicated hardware before 
you, so I didn't know.
- I'm just not very rich and try to integrate some things I need in my 
machine (don't worry, I did not framebuffered or X.orged it yet) because 
I cannot afford to buy another one (yes, even the 200€ one)... The part 
you don't want to know is how many people I had to kill in order to get 
my TDM400 card, until I found out that other cheaper solutions existed. :-)

 We're just trying to help -- but if you insist on running Asterisk in a
 VM, then you're on your own.
   
And I thank you for that (the helping part), you've found the deep cause 
of all my zaptel problems (Xen), so please don't leave me alone! ;-)

To be a bit more constructive, I'd like to ask you or anyone that dared 
to try using Asterisk on a non-dedicated hardware, specifically those 
that tried on a machine hosting VMs the following:

- If there is no way running Asterisk with Xen, what type of 
'hypervisor' should I use in order not to have problems? KVM?, KQemu?, 
VMWare?
- What type of problems should I expect if I dare to do that? (of 
course, Asterisk will be realtime-niced to make it more important)


Thanks and sorry again for the misunderstandings,
François.
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RE: [asterisk-users] [*Win32 0.60] Sending call notification bye-mail/web?

[Cosmin Prund]

I'm using an FastAGI written in Delphi for my IVR so I can confirm it
works just fine. I wrote all the code from scratch and it wasn't a big
deal, but you can find sample code on Free Pascal sites (google will
help you).

Also I'd recommend turning your idea into an FastAGI. It will work with
both native (Linux) Asterisk and with the Win32 port, and it will
actually be easier to debug! You just start your FastAGI server exe,
place a brakepoint in the code, pick up your phone and dial your test
number. Asterisk has long-enough timeouts when talking to an FastAGI
application to make stepping through the code possible.

--
Cosmin Prund

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nick
Seraphin
Sent: Tuesday, May 15, 2007 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [*Win32 0.60] Sending call notification
bye-mail/web?


On Tue, 15 May 2007, Vincent Delporte wrote:

 Hello,
 
 In case there are other users of the AsteriskWin32 port...
 
 I haven't really used the AGI feature of Asterisk to run an
application 
 from extensions.conf. *Win32 supports Perl, which I don't know.
Apparently, 
 it's also possible to write AGI applications as EXE's (there's a 
 eagi-test.exe file installed by default).
 
 = When a call comes in, I'd like an AGI application to send an e-mail
and 
 send CID name/number to a script on a web server.
 
 Is this the correct way to do it in Perl, with the modules available
in 
 AsteriskWin32? Could I rewrite this in Delphi instead?


ALL AGI scripts are basically just programs that read from stdin and
write
to stdout.  They can therefore be written in almost any language.  So
yes,
Delphi should work fine.

(I have very fond memories of Delphi, and before that, Borland Pascal w/
Objects for DOS, and before that, Turbo Pascal...  one of these days
I'll
have to get the latest version of Delphi and take a walk down memory
lane.
These days everything is C this or Perl that.  I loved Pascal. :-)) 

-- Nick


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RE: [asterisk-users] Linksys SPA3012 inbound FXO problems

[Cosmin Prund]

I've managed to configure a SPA3012 to do that a few days ago. I
remamber using something like S0:[EMAIL PROTECTED] for the
#1 dial plan. Unfortunately I no longer have access to the SPA because I
shiped it to an co-worker and this co-worker didn't manage to install
it yet.

I also remamber an odd thing: the extension really needs to exist in the
correct context, it doesn't fall back to the s extension and there's
no worning on the CLI ither!

Also an googling tip: most configuration for the SPA3012 is the same as
that for SIPURA 3000, so google for that too. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: Thursday, May 03, 2007 6:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Linksys SPA3012 inbound FXO problems

Hello list,
hope someone can help me - I'm going crazy using the FXO port a SPA3012.
I would like the SPA 3012 to act as a simple FXO port to an Asterisk,
that  
is, once it detects a call, it should simply send it over to the local  
Asterisk server. No intelligent routing, PIN, anything else

I configured it like this:

PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no
PSTN Caller Default DP: 1

Then I configured the dialplan #1 as:
Dial Plan 1: (S0:@gw1)

And I configured gateway 1 as:

Gateway Accounts
Gateway 1: my.asterisk.server   
GW1 NAT Mapping Enable:  no
GW1 Auth ID:   --my-sip-login--
GW1 Password:  --my-sip-password--

But it seems to simply ignore incoming calls at all
Anybody's got a pointer to get me started?
Thanks in advance,
l.



-- 
Home of QueueMetrics - http://queuemetrics.com

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RE: [asterisk-users] Asterisk-1.4.3

[Cosmin Prund]

Are you using automatic rules for sorting email?

My Outlook 2007 miss-sorted all of those announcements (for Asterisk 1.4.3 and 
a few others - that's the only one I cared about) and I missed them. Maybe your 
mail roules are just as wrong.

--
Cosmin Prund

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon
Sent: Wednesday, April 25, 2007 5:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk-1.4.3

Richard Klingler wrote:
 Hello (o;
 
 
 Did I miss somewhere the announcement of 1.4.3?
 Also don't see anything in the announce mailing
 list archive...but it is available for download...
 
Also didn't spot zaptel 1.4.2, weird. (I read the security announcement
and was silly enough to assume that although it stated the fixes were in
1.4.3, that they would only be available by CVS at the time).
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RE: [asterisk-users] asterisk answering machine

[Cosmin Prund]

If you're learning Asterisk right now, you might try using basic
Dialplan first, so less things may go wrong. There's a dialplan function
that does what you want:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GotoIfTime
As for white-listing CallerId's, you may use simple GotoIf's in the
Dialplan, or you may use an AGI to consult a CallerId database (plain
text file). 

On the other hand, if you've got your AGI going, stick to that!

--
Cosmin Prund 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Troy Ayers
Sent: Thursday, April 26, 2007 3:59 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk answering machine

I'm learning asterisk, and decided to make myself an answering machine 
out of it.  Seems pretty straightforward to use an agi (perl) to do what

I want.

What I want is:
Answer the phone.
check for time of the day
If TOD is during the time I sleep I announce i'm sleeping  prompt 
caller to dial1 (or whatever) to connect to my extension  then go to 
voicemail if busy/una, otherwise go straight to voicemail.if no digit 
was pressed.

If TOD is during normal waking hours or caller ID matches whitelisted 
numbers, just connect to my extension  then go to voicemail if
busy/una.

I'm nearly done, but I had a thought: before I re-invent the wheel, does

anyone know if this has already been done?  My searches only saw basic 
answering machines examples.

-Troy


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RE: [asterisk-users] analog line cards / adapter

[Cosmin Prund]

How about connecting the analog phone in the elevator to an ATA gateway
that provides PSTN FallThrow? Sipura 3000 or Lynksis SPA3102 can do
the trick.

--
Cosmin Prund

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik
Wartusch
Sent: Thursday, April 26, 2007 12:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] analog line cards / adapter

Hi,

Is anybody aware about a device (e.g. PCI card) that can handle a analog
phone 
for Asterisk and can loop through the line directly in case of an power
fault 
in the server. Its for an emergency phone in an elevator, so if the
power is 
down the phone has to have the possibilite to make a call outside (and 
powered of course over the POTS net), in normal operation it should be 
connected to Asterisk and the Server. Some suggestions or running 
environments?
Thanks
Kind Regards
Erik
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RE: [asterisk-users] Improve voice quality on Asterisk +chan_capi+DIVABRI

[Cosmin Prund]

Thanks a lot, that was it! I had both softdmtf=on and relasdtmf=on.
I only touched softdmtf now, but I might have played with relaxdtfm
before this. It now works fine with DTMF clamping activated.


 Both logs don't show any DTMF activity. DMTF detection is not
activated
 at
 all. Please make sure you DON'T have softdmtf=yes or relaxdtmf=yes in
 your
 capi.conf.

--
Thanks,
Cosmin Prund
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RE: [asterisk-users] Improve voice quality on Asterisk +chan_capi+DIVA BRI

[Cosmin Prund]

-activare'
(language 'de')
CONNECT_ACTIVE_IND ID=001 #0x4a98 LEN=0015
  Controller/PLCI/NCCI= 0x301
  ConnectedNumber = default
  ConnectedSubaddress = default
  LLC = default

CONNECT_ACTIVE_RESP ID=001 #0x4a98 LEN=0012
  Controller/PLCI/NCCI= 0x301

CONNECT_B3_IND ID=001 #0x4a99 LEN=0013
  Controller/PLCI/NCCI= 0x2c0301
  NCPI= default

CONNECT_B3_RESP ID=001 #0x4a99 LEN=0015
  Controller/PLCI/NCCI= 0x2c0301
  Reject  = 0x0
  NCPI= default

CONNECT_B3_ACTIVE_IND ID=001 #0x4a9a LEN=0013
  Controller/PLCI/NCCI= 0x2c0301
  NCPI= default

CONNECT_B3_ACTIVE_RESP ID=001 #0x4a9a LEN=0012
  Controller/PLCI/NCCI= 0x2c0301

  == ISDN1#02: Setting up echo canceller (PLCI=0x301, function=1,
options=4, tail=0)
FACILITY_REQ ID=001 #0x363c LEN=0024
  Controller/PLCI/NCCI= 0x301
  FacilitySelector= 0x8
  FacilityRequestParameter= 01 00 06 04 00 00 00 00 00

FACILITY_CONF ID=001 #0x363c LEN=0022
  Controller/PLCI/NCCI= 0x301
  Info= 0x0
  FacilitySelector= 0x8
  FacilityConfirmationParameter   = 01 00 02 00 00

-- ISDN1#02: Echo canceller successfully set up (PLCI=0x301)
INFO_IND ID=001 #0x4bbc LEN=0017
  Controller/PLCI/NCCI= 0x301
  InfoNumber  = 0x1e
  InfoElement = 82 88

INFO_RESP ID=001 #0x4bbc LEN=0012
  Controller/PLCI/NCCI= 0x301

-- ISDN1#02: info element PI 82 88
ISDN1#02: In-band information available
INFO_IND ID=001 #0x4bbd LEN=0017
  Controller/PLCI/NCCI= 0x301
  InfoNumber  = 0x1e
  InfoElement = 82 83

INFO_RESP ID=001 #0x4bbd LEN=0012
  Controller/PLCI/NCCI= 0x301

-- ISDN1#02: info element PI 82 83
ISDN1#02: Origination is non ISDN
INFO_IND ID=001 #0x4bbe LEN=0017
  Controller/PLCI/NCCI= 0x301
  InfoNumber  = 0x8
  InfoElement = 80 90

INFO_RESP ID=001 #0x4bbe LEN=0012
  Controller/PLCI/NCCI= 0x301

-- ISDN1#02: info element CAUSE 80 90
INFO_IND ID=001 #0x4bbf LEN=0015
  Controller/PLCI/NCCI= 0x301
  InfoNumber  = 0x8045
  InfoElement = default

INFO_RESP ID=001 #0x4bbf LEN=0012
  Controller/PLCI/NCCI= 0x301

-- ISDN1#02: info element DISCONNECT
-- ISDN1#02: Disconnect case 3
-- CAPI queue frame: TYPE: Control (4) SUBCLASS: Hangup (1) ]
[ISDN1#02]
/CLI Output


--
Thanks,
Cosmin Prund



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Armin Schindler
 Sent: 20 aprilie 2007 14:48
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Improve voice quality on Asterisk
 +chan_capi+DIVA BRI
 
 On Fri, 20 Apr 2007, Cosmin Prund wrote:
  I've implemented my IVR using an FastAGI thing, using the READ
  application. core show application read shows no information on
how
  the read function gets it's digits, I assume it does it the right
 way.
  With DTMF clamping off it works, with DTMF clamping on it no longer
  works. I've also toggled the softftfm setting in capi.conf, no
luck
  ether way.
 
  Is there anything else I can try? Did I miss the obvious (it would
 not
  be my first)
 
 Can you please create a capi log:
   set verbose 5
   capi debug
 to see what really happens via the interface?
 
 Armin
 
   -Original Message-
   From: [EMAIL PROTECTED] [mailto:asterisk-
 users-
   [EMAIL PROTECTED] On Behalf Of Armin Schindler
   Sent: 20 aprilie 2007 12:32
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: RE: [asterisk-users] Improve voice quality on Asterisk +
   chan_capi+DIVA BRI
  
   On Fri, 20 Apr 2007, Cosmin Prund wrote:
Ok, I've made all those changes, called my operator from an
 outside
   line
and tried alternatively whispering / shouting into the mic,
 banging
   the
microphone with a metal object and pressing DTMF digits.
   
So far - so good, it seems to work.
   
I've now got an other problem. Clamping DTMF disabled my IVR! Is
   there
any way to enable/disable DTMF clamping on a per-call basis? Or
   better,
disable DTMF only when the call makes it to an operator?
  
   This is possible, but such a command/feature must be implemented
 into
   chan-capi first.
   Anyway, even with DTMF clamping the DTMF detection is activated.
So
   Asterisk should get the DTMF infos. Or is your IVR doing own DTMF
   detection on voice data? If yes, you should change that.
  
   Armin
  
 -Original Message-
 From: [EMAIL PROTECTED]
[mailto:asterisk-
   users-
 [EMAIL PROTECTED] On Behalf Of Armin Schindler
 Sent: 19

RE: [asterisk-users] Improve voice quality on Asterisk + chan_capi+ DIVA BRI

[Cosmin Prund]

Ok, I've made all those changes, called my operator from an outside line
and tried alternatively whispering / shouting into the mic, banging the
microphone with a metal object and pressing DTMF digits.

So far - so good, it seems to work.

I've now got an other problem. Clamping DTMF disabled my IVR! Is there
any way to enable/disable DTMF clamping on a per-call basis? Or better,
disable DTMF only when the call makes it to an operator?

--
Thanks,
Cosmin Prund

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Armin Schindler
 Sent: 19 aprilie 2007 14:35
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Improve voice quality on Asterisk +
 chan_capi+ DIVA BRI
 
 On Thu, 19 Apr 2007, Cosmin Prund wrote:
  Hello everyone!
 
  I've got a Eicon Diva Server BRI card into my * box working just
 fine,
  but I wander if there's anything I can do to improve voice quality
 for
  my operators. I'm thinking something along the lines of auto gain
 and
  sudden noise suppression (like when you hit a fax machine or the
 other
  party accidently touches the dial pad on the phone).
 
  Does one of Asterisk, chan_capi or the Diva driver have support for
 such
  functionality?
 
 Sure, with the Dialogic (Eicon) DIVA Server card DSPs, you have the
 following possibilities:
 
 1. Automatic Gain Control and Active Talker Evaluation in conference
 (by
default automatically activated with three or more parties)
 2. Recording Stream Automatic Gain Control
 3. Manual Control of Signal Level
 4. Manual control of the signal pitch and/or bitrate (rate conversion)
 5. Suppression of DTMF tones. This feature can be activated using
 adapter
configuration (for all calls) or on per call basis
This is always good to activate this feature for operators to
 protect
people from signals or in one gateway to prevent DTMF tones from
 passing
through gateway in band.
The DTMF tones are suppressed in the way which will not affect the
quality of the voice signal in case voice signal and DTMF tones
 overlap.
 6. Part 68 Voice Signal Limiter (Required in US, by default
deactivated
 in
Europe). This protects the ears from clicks and too loud signals.
 This
feature can be activated using the configuration. This is good idea
 to
activate Part 68 voice signal limiter to protect the people. This
is
 the
dynamic voice signal limiter in accordance with Part 68 of US
requirements.
 
 The Part 68 Limiter, Audio Recording Automatic Gain Control (AGC of
 received signal) and the DTMF Clamping (Suppression of DTMF tones) are
 can be controlled using adapter configuration and do not require any
 change in the application (but can be controlled on the per call basis
 too, which is not implemented in chan-capi yet).
 
 
 Armin
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RE: [asterisk-users] Improve voice quality on Asterisk + chan_capi+DIVA BRI

[Cosmin Prund]

I've implemented my IVR using an FastAGI thing, using the READ
application. core show application read shows no information on how
the read function gets it's digits, I assume it does it the right way.
With DTMF clamping off it works, with DTMF clamping on it no longer
works. I've also toggled the softftfm setting in capi.conf, no luck
ether way.

Is there anything else I can try? Did I miss the obvious (it would not
be my first)

--
Thanks,
Cosmin Prund



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Armin Schindler
 Sent: 20 aprilie 2007 12:32
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Improve voice quality on Asterisk +
 chan_capi+DIVA BRI
 
 On Fri, 20 Apr 2007, Cosmin Prund wrote:
  Ok, I've made all those changes, called my operator from an outside
 line
  and tried alternatively whispering / shouting into the mic, banging
 the
  microphone with a metal object and pressing DTMF digits.
 
  So far - so good, it seems to work.
 
  I've now got an other problem. Clamping DTMF disabled my IVR! Is
 there
  any way to enable/disable DTMF clamping on a per-call basis? Or
 better,
  disable DTMF only when the call makes it to an operator?
 
 This is possible, but such a command/feature must be implemented into
 chan-capi first.
 Anyway, even with DTMF clamping the DTMF detection is activated. So
 Asterisk should get the DTMF infos. Or is your IVR doing own DTMF
 detection on voice data? If yes, you should change that.
 
 Armin
 
   -Original Message-
   From: [EMAIL PROTECTED] [mailto:asterisk-
 users-
   [EMAIL PROTECTED] On Behalf Of Armin Schindler
   Sent: 19 aprilie 2007 14:35
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] Improve voice quality on Asterisk +
   chan_capi+ DIVA BRI
  
   On Thu, 19 Apr 2007, Cosmin Prund wrote:
Hello everyone!
   
I've got a Eicon Diva Server BRI card into my * box working
 just
   fine,
but I wander if there's anything I can do to improve voice
 quality
   for
my operators. I'm thinking something along the lines of auto
 gain
   and
sudden noise suppression (like when you hit a fax machine or the
   other
party accidently touches the dial pad on the phone).
   
Does one of Asterisk, chan_capi or the Diva driver have support
 for
   such
functionality?
  
   Sure, with the Dialogic (Eicon) DIVA Server card DSPs, you have
the
   following possibilities:
  
   1. Automatic Gain Control and Active Talker Evaluation in
 conference
   (by
  default automatically activated with three or more parties)
   2. Recording Stream Automatic Gain Control
   3. Manual Control of Signal Level
   4. Manual control of the signal pitch and/or bitrate (rate
 conversion)
   5. Suppression of DTMF tones. This feature can be activated using
   adapter
  configuration (for all calls) or on per call basis
  This is always good to activate this feature for operators to
   protect
  people from signals or in one gateway to prevent DTMF tones
from
   passing
  through gateway in band.
  The DTMF tones are suppressed in the way which will not affect
 the
  quality of the voice signal in case voice signal and DTMF tones
   overlap.
   6. Part 68 Voice Signal Limiter (Required in US, by default
  deactivated
   in
  Europe). This protects the ears from clicks and too loud
 signals.
   This
  feature can be activated using the configuration. This is good
 idea
   to
  activate Part 68 voice signal limiter to protect the people.
 This
  is
   the
  dynamic voice signal limiter in accordance with Part 68 of US
  requirements.
  
   The Part 68 Limiter, Audio Recording Automatic Gain Control (AGC
of
   received signal) and the DTMF Clamping (Suppression of DTMF tones)
 are
   can be controlled using adapter configuration and do not require
 any
   change in the application (but can be controlled on the per call
 basis
   too, which is not implemented in chan-capi yet).
  
  
   Armin
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[asterisk-users] Improve voice quality on Asterisk + chan_capi + DIVA BRI

[Cosmin Prund]

Hello everyone!

 

I've got a Eicon Diva Server BRI card into my * box working just fine,
but I wander if there's anything I can do to improve voice quality for
my operators. I'm thinking something along the lines of auto gain and
sudden noise suppression (like when you hit a fax machine or the other
party accidently touches the dial pad on the phone).

 

Does one of Asterisk, chan_capi or the Diva driver have support for such
functionality?

 

 

--

Thanks,

Cosmin Prund

 

 

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Re: [asterisk-users] Playback 5% Too Fast?

[Cosmin Prund]

I've had similar behavior on my own IVR. I moved my sound files to a ram 
disk and all pops and ticks stopped!


David Brazier wrote:

Hi All

I have a problem with IVR scripts which consist mainly of Playback of
audio files, driven from an AGI application.  There are clicks every few
seconds or more frequently that is audible on the remote end (PSTN), but
not on the Asterisk recording of the call.  If I record the remote end
and compare it to the local recording, it appears to be about 5%-7% too
fast - i.e. if I synchronise the starts, the remote end finishes sooner.
I can find points in the remote recording where parts of the waveform
have been missed out, leading to jumps in the waveform, which correspond
to the audible clicks.  These jumps seem like dropped packets, and I'm
deducing that Asterisk is sending data slightly too fast (i.e. more
frequently than 50x160 sample per second) for the remote end, which has
to drop data to keep up.  


This is a VoIP-only set up - no Zap hardware.  Thinking this was a
timing issue, I have installed Zaptel to get ztdummy, which is loaded
OK, but that hasn't made any difference.  I have tried it with different
VoIP providers and observed the same problem.

Behaviour has persisted from 1.2 to 1.4 and now 1.4.1.  CentOS 4.4
(2.6.9 kernel), Dell 1950.

Any ideas how to progress?  Is this a timing issue or am I wide of the
mark?

Thanks for any help

David
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Re: [asterisk-users] gtalktovoip and Asteirsk

[Cosmin Prund]

I don't think it works. I tried calling my own yahoo messenger ID with 
no success: it rings a number of times and then it goes to some sort of 
voice mail.
And I did invite the user they specified to my yahoo list, I also 
entered my yahoo id into the registration form on the site.

I used a extensions.conf command like this for the try:

exten = 641,1,Dial(SIP/[EMAIL PROTECTED])

(and yes, that's one of the yahoo ID I tryed with, and I don't think it 
exists! )


Klaverstyn, David C wrote:


Has anyone managed to get gtalktovoip working at all?  If so please 
explain.


 


http://www.gtalk2voip.com/faq.shtml

 

 


*2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ?*

A: This is a major feature of our gateway and it is very easy.

oGTalk: [EMAIL PROTECTED] can be reached by calling to 
sip:[EMAIL PROTECTED]


oMSN: [EMAIL PROTECTED] can be reached by calling to 
sip:[EMAIL PROTECTED]


oYahoo: [EMAIL PROTECTED] can be reached by calling to 
sip:[EMAIL PROTECTED]


 




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Re: [asterisk-users] Billing pulses

[Cosmin Prund]

The network terminator installed by the Telco in Romania works the same 
way: it has two analog outputs and two digital (S0) outputs. I've also 
got a full TDM400 card with 3 FXS and one FXO, but I gave them up gladly 
for a proper ISDN card (I'm using a Diva Eicon Server) - and I don't do 
billing. Sound quality is perfect, there's no echo and I can use all the 
functions of the ISDN card, like the ability to use multiple MSN's, send 
an proper busy signal at will, get two calls on the same number at the 
same time.


And now I've got two unused FXS ports in my Asterisk.

Stefano Corsi wrote:
I must clarify my original message. Maybe confusion is due to my poor 
english. So I'll make a list of statements:


- Each ISDN line in Italy can be splitted in two analog lines
- You can use those analog lines as normal analog lines
- I have already invested in analog hardware (my fault of course) for 
both FSX and FXO
- ISDN hardware installed by the telco can, in Italy, be programmed to 
send a billing pulse.
- I guess this billing pulse is sent on each of the two analog lines 
in which a single ISDN line can be splitted (so there's no risk, I 
guess, for double billing).
- I'm considering if there's a small chance for me to avoid buying 
additional hardware (ISDN cards or gateways) and have an accurate 
billing using those analog lines resulting from splitting an ISDN line.
- To get an accurate billing, I'm wandering if it's possibile to use 
billing pulse provided by those analog lines.

- I have full specifications of the billing pulse provided:

frequency 
 
12 kHz ± 1%
level 
.. 
200 mVrms on 200
distortion... 
 5%
pulse duration 
.125 ± 25 ms
pause duration 
 180 ms
period 
... 
300 ms


Do you think it's worth considering it?

Rgds
Stefano

 Bill them both.  We are talking about mere BRI's, right:-)  Good 
catch,
 David.  As others noted, billing pulse really applies to analogue 
lines

 only, and ISDN providers should always send status.

 Yuan Liu

Thanks, Yuan


But my confusion came from the original post stating the use of ISDN
circuits for this  implementation.  Id ISDN is in fact the circuit of
choice for this app, I agree why wouldn't he simply use the cause codes
for billing purposes.  We have a lot of experience in telecommunications
billing, and have always found cause codes to be more than sufficient
even for weird tiers, and bizarre rounding functions.



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Re: [asterisk-users] Trying to register an G.729 codec boght from Digium and the register command does aboslutely nothing

[Cosmin Prund]

Digium support cleared the issue for me, they sent me a new register 
utiliy by mail and this one worked as expected. I registered my codedc 
and tested my codec. If anyone needs to know, I tested the codec using a 
SIPURA 3000 ATA so I can confirm this ATA works with G729.


I'd like to add: Digium support responded very quickly.

Cosmin Prund wrote:
Hello: I got into a trap. As far as I know I do not need to pay any 
royalties to use G.729b in Romania, so I should have used other 
drivers. The installation procedure looked difficult so I decided to 
get one from Digium - it's not that expensive, my time is much more 
expensive.


Made the payment, got they key, downloaded and copied everything as in 
http://kb.digium.com/entry/30/5/; but when I called register I got 
no result. Actually I do get the prompt asking me to use -l to see 
the licence, nothing after that. It gives no error message, nothing at 
all!


My first ethernet device is eth0 so it's not that; I'm able to 
browse https sites so the ports are open. I *disabled* the firewall 
and tried again, no success, so it's not firewall related. What to do 
next?


Thanks.
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Re: [asterisk-users] New Issue

[Cosmin Prund]

the ./configure thing requires the sources of zaptel, not asterisk. 
Are you sure they're passing the zaptel sources?


Well... i'm out of ideas. If it doesn't work you might want to re-post 
your thread (specifically say you don't see chan_zap in make menuconfig) 
and start with a new message (send) - don't reply to an existing message 
and change it's subject line. When you first posted this message you 
hijacked a thread called Mysterious tables starting with stats_. 
People using threaded mail readers might not even see your question! I 
saw your question because the thread about Mysterious stats_ tables 
looked interesting...


David Ruggles wrote:

I'm still not seeing chan_zap in menu option three.

I copied the source directories from /root/downloads/asterisk (where I had
put them) to /usr/src/ and then did what you suggested below and I got the
same result.

I'm going to try make uninstalling all the packages deleted all source
directories and starting over from the downloads. If you any other
suggestions I'll do them.

TIA!!

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund
Sent: Tuesday, February 06, 2007 6:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] New Issue


Try it like this:

cd /usr/src/asterisk-1.4.0
make clean
./configure --with-zaptel=/usr/src/zaptel-1.4
make menuconfig
make all
make install

David Ruggles wrote:
  

Sorry about that I must have been in the wrong directory. I also have


1.4.0
  

and I tried it again and it worked. Chan_zap is not listed there, I'll


start
  

poking around and see if I stumble across anything. Do you know where the
expected location is? I don't have a problem moving the source.

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]
  



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[asterisk-users] Trying to register an G.729 codec boght from Digium and the register command does aboslutely nothing

[Cosmin Prund]

Hello: I got into a trap. As far as I know I do not need to pay any 
royalties to use G.729b in Romania, so I should have used other drivers. 
The installation procedure looked difficult so I decided to get one from 
Digium - it's not that expensive, my time is much more expensive.


Made the payment, got they key, downloaded and copied everything as in 
http://kb.digium.com/entry/30/5/; but when I called register I got no 
result. Actually I do get the prompt asking me to use -l to see the 
licence, nothing after that. It gives no error message, nothing at all!


My first ethernet device is eth0 so it's not that; I'm able to browse 
https sites so the ports are open. I *disabled* the firewall and tried 
again, no success, so it's not firewall related. What to do next?


Thanks.
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Re: [asterisk-users] Trying to register an G.729 codec boght from Digium and the register command does aboslutely nothing

[Cosmin Prund]

So simple... I'm doing that right now, I've sent them an email.
I didn't find that email address on Digium's support page...

Thanks.

Bruce Ferrell wrote:



Cosmin Prund wrote:
Hello: I got into a trap. As far as I know I do not need to pay any 
royalties to use G.729b in Romania, so I should have used other 
drivers. The installation procedure looked difficult so I decided to 
get one from Digium - it's not that expensive, my time is much more 
expensive.


Made the payment, got they key, downloaded and copied everything as 
in http://kb.digium.com/entry/30/5/; but when I called register I 
got no result. Actually I do get the prompt asking me to use -l to 
see the licence, nothing after that. It gives no error message, 
nothing at all!


My first ethernet device is eth0 so it's not that; I'm able to 
browse https sites so the ports are open. I *disabled* the firewall 
and tried again, no success, so it's not firewall related. What to do 
next?


Thanks.
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Contact [EMAIL PROTECTED]



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Re: [asterisk-users] New Issue

[Cosmin Prund]

Let's see if I remember this, it gave me a bit of trouble as well.
*after* you made sure you've got the zaptel driver in order, go to the 
src folder for asterysk and issue make menuconfig.
Go to 3 and see if you have the chan_zap listed there and with [*] 
prefix. If it's not listed it's because you've got the zaptel driver 
sources in an unexpected location. You'll need to manually specify the 
location of you zaptel driver (don't remember how) and then re-issue the 
make menuconfig. This time you'll see the zap chan driver available as 
an option at 3. Now exist the make menuconfig SAVING your changes. I 
didn't save (since I didn't make any changes) and my channel driver 
didn't build. I tried it again, saved the config and it worked.


Hope someone can fell the missing bit, the way to tell make where to 
find the zaptel source files.


David Ruggles wrote:

Well that didn't work. I still don't have a zap channel driver. What else
can I try?

TIA!

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles
Sent: Tuesday, February 06, 2007 4:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] New Issue


I'm missing chan_zap.so, I'm going to make and make install again as per:
http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles
Sent: Tuesday, February 06, 2007 3:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] New Issue


Now that ztcfg is working correctly I can't seem to get asterisk to answer a
call.

I did the make install and make samples so I have a lot of configuration
files that I know nothing about.

Here is contents of zapata.conf
[trunkgroups]

[channels]
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
channel = 1 


And the contents of extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest   ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1  ; MSD digits to strip
(usually 1 or 0)

[incoming]
exten = s,1,Answer()
exten = s,2,Echo()

This from TFOT, the general and globals sections of extensions came from the
sample. I started Asterisk using asterisk -cvvv and it doesn't seem to have
any errors, but I can't find where it parses zapata.conf. I do see it
parsing extensions.conf

What should I do?

TIA!!

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]


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Re: [asterisk-users] New Issue

[Cosmin Prund]

I'v got Asterisk 1.4.0 and it understands make menuconfig. Is your 
version older or newer? If it's older, maybe you can try the newer one. 
If it's newer - I'm out of ideas.


David Ruggles wrote:

Thanks for the reply, but when I go to the asterisk source directory and
issue make menuconfig I get:
make: *** No rule to make target `menuconfig'. Stop.

The source I have is the latest tar file from the astrisk site.
Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]
  


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Re: [asterisk-users] New Issue

[Cosmin Prund]

Try it like this:

cd /usr/src/asterisk-1.4.0
make clean
./configure --with-zaptel=/usr/src/zaptel-1.4
make menuconfig
make all
make install

David Ruggles wrote:

Sorry about that I must have been in the wrong directory. I also have 1.4.0
and I tried it again and it worked. Chan_zap is not listed there, I'll start
poking around and see if I stumble across anything. Do you know where the
expected location is? I don't have a problem moving the source.

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  [EMAIL PROTECTED]
  


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Re: [asterisk-users] Give Busy to the 3rd call on a BRI using chan_capi

[Cosmin Prund]

Thanks, it really was easy.
Unfortunately it only works for MSN's, not for the base number. Oh 
well, I'll just stop using the base number, I've got enough MSN's anyway.


Thanks again.

Armin Schindler wrote:

On Thu, 1 Feb 2007, Cosmin Prund wrote:
  

Any ideas? It should be simple...



It is easy: read the README in chan-capi.org package ;-)

Just look into the variable BCHANNELINFO and you will know if it is a call
without b-channel (the third call).

Armin
  
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Re: [asterisk-users] Give Busy to the 3rd call on a BRI using chan_capi

[Cosmin Prund]

Armin Schindler wrote:
What type of line is that? The 'base number' is also a MSN on lines I know. 
Or is it PtP with DID?


Armin

On Sat, 3 Feb 2007, Cosmin Prund wrote:


The base number works like any other MSN most of the times, but the 
busy application doesn't work on it. If I dial the base number while 
there are 2 calls in progress, I can see the busy application being 
called (in the CLI) but I still here ringing on the test phone.


Doing something really easy like:

[capi-in]
exten = base_number,1,Busy
exten = msn,1,Busy

will work for the msn but not for the base_number. I don't think this is 
a chan_capi issue, I think this is a Telco issue. The telco treats the 
base number differently on other issues as well.


As for what kind of line is that - well - guess is the msn kind. The 
telko is refering to the extra numbers as msns.

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Re: [asterisk-users] Give Busy to the 3rd call on a BRI using chan_capi

[Cosmin Prund]

Tzafrir Cohen wrote:

Do you use Busy to send a bus signal to the other party?

  


I use Busy. I have no idea how it works. When I call from my mobile 
phone to my PBX I get a busy signal and it seems I'm not being charged 
for the call (so it's not like * opened up the line and played the 
busy signal). It also works if I call from an other land line.

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Re: [asterisk-users] Give Busy to the 3rd call on a BRI using chan_capi

[Cosmin Prund]

Any ideas? It should be simple...

Cosmin Prund wrote:

Hello everyone:

using chan_capi 0.7 and asterisk 1.4

Quick question:
How can I detect the number of voice channels (B channels) in use at 
a given time. I'd like to call Busy if two B channels are used on my 
BRI to give the calling customer a Busy signal.


Long question:
On my single-line BRI (two channels) I'd like to give the 3rd 
simultaneous incoming call an busy signal. I already tested and the 
Busy function works very well (I've set up one of my MSN's to 
immediately call Busy). I also tested and I'm 100% sure the 3rd call 
makes it into the box while the other 2 channels are talking, so this 
is not a Telco problem and it can be fixed locally. Doing this on my 
side of the line (as opposed to having the Telco issue the Busy signal 
on my behalf) has an number of benefits: (a) I don't need to talk to 
the Telco (b) I *know* who called and I can call them back and (c) In 
a distant future I might use the capi channel's ability to transfer 
the call to a different POTS line since this doesn't use the B channel.


Thanks,
Cosmin Prund
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[asterisk-users] Give Busy to the 3rd call on a BRI using chan_capi

[Cosmin Prund]

Hello everyone:

using chan_capi 0.7 and asterisk 1.4

Quick question:
How can I detect the number of voice channels (B channels) in use at a 
given time. I'd like to call Busy if two B channels are used on my BRI 
to give the calling customer a Busy signal.


Long question:
On my single-line BRI (two channels) I'd like to give the 3rd 
simultaneous incoming call an busy signal. I already tested and the Busy 
function works very well (I've set up one of my MSN's to immediately 
call Busy). I also tested and I'm 100% sure the 3rd call makes it into 
the box while the other 2 channels are talking, so this is not a Telco 
problem and it can be fixed locally. Doing this on my side of the line 
(as opposed to having the Telco issue the Busy signal on my behalf) has 
an number of benefits: (a) I don't need to talk to the Telco (b) I 
*know* who called and I can call them back and (c) In a distant future I 
might use the capi channel's ability to transfer the call to a different 
POTS line since this doesn't use the B channel.


Thanks,
Cosmin Prund
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[asterisk-users] Mabe OT? What managed switch is best for VoIP application?

[Cosmin Prund]

My Trendnet 26 port managed switch gave up on me so I'm shopping for a 
new switch. I learned the hard way NOT to trust marketing material from 
anyone so now I'm asking the list: what am I looking for in a managed, 
VoIP switch?


P.S: For those that don't understand WHY I can't trust marketing 
material, let me tell you something about the Trendnet switch that's 
fast becoming garbidge. I wanted an managed switch so I boght the 
switch had Managed and Virtual LAN in the biggest possible letters. 
Later, after buying two Intel 1Gb Virtual Lan Enabled network cards, I 
discovered my Trendnet switch doesn't do standard VLan, it only does 
VLan if linked to an other Trendnet switch - not useful at all!


Thanks,
Cosmin Prund
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Re: [asterisk-users] Mabe OT? What managed switch is best for VoIP application?

[Cosmin Prund]

No one told me about 802.1q Vlan before I boght the switch. It was 
printed in big fat letters on the box. Now I *do* know about 802.1q but 
it's a little bit too late: I already have the switch. Fortunately 
(unfortunately) the switch is gone, it's dead. Now I want a better 
switch and I'm asking so I don't fall into the same trap *again*.


Again, this big switch is not the only device I bought only to find out 
it doesn't exactly do what I want it to do. I also got a nice little 
ZyXEL VPN collecting dust in a drawer somewhere. I wanted a VPN 
router/firewall that allowed me to connect to my network from my 
Windows-based Laptop computer, using the tools available in the system. 
Guess what: I *can* connect to the ZyXEL using an paid-for client that 
costs almost as much as the firewall itself. I'm now running PopTop on 
my Linux Asterisk box and it works just fine, and it's a lot cheaper. 
And I did learn about a few other standards names in the process: AFTER 
I bought the hardware device.


So the idea is very simple: I need a switch that does VoIP well, has 
lots of ports and does 802.1q VLAN. I also want it to be managed and 
have it's management tools help me diagnose problems. That's my biggest 
question right now: What *exactly* am I looking for? My Trendent switch 
has management and it's easy to use for what it does, but it would never 
help me diagnose a network problem. It took a number of disconected 
*local* LAN VoIP calls before I noticed the switch is flowed and needs 
to be replaced.


Thanks,
Cosmin Prund

Patrick Cervicek wrote:

Cosmin Prund schrieb:

P.S: For those that don't understand WHY I can't trust marketing 
material, let me tell you something about the Trendnet switch that's 
fast becoming garbidge. I wanted an managed switch so I boght the 
switch had Managed and Virtual LAN in the biggest possible 
letters. Later, after buying two Intel 1Gb Virtual Lan Enabled 
network cards, I discovered my Trendnet switch doesn't do standard 
VLan, it only does VLan if linked to an other Trendnet switch - not 
useful at all!


Standard Vlan = 802.1q

Trendnet offered you only VLAN in the Switch, not 802.1q

You have to look for the Protocol *802.1q*
http://en.wikipedia.org/wiki/VLAN#Protocols_and_design

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[asterisk-users] Failing to compile chan_capi

[Cosmin Prund]

I've got a brand new Eicon Diva Server BRI card and I want to configure 
it with Asterisk. I managed to get asterisk and zaptel to compile and 
install, I've compiled and installed the drivers for the Diva card and 
now I need to compile and install the chan_driver for chan_capi. 
Unfortunately this fails miserably. I get the following messages:


I'm using: Kernel 2.6.16.37.4, zaptel-1.4.0 and asterisk-1.4.0., 
chan_capi 0.7.1


//--

[EMAIL PROTECTED] chan_capi-0.7.1]# make
./create_config.sh /usr/src/asterisk-1.4.0/include
Checking Asterisk version... 1.4.0
* found stringfield in ast_channel
* found 'indicate' with data
config.h complete.
gcc -pipe -fPIC -Wall -Wmissing-prototypes -Wmissing-declarations -g  
-I/usr/src/asterisk-1.4.0/include -D_REENTRANT -D_GNU_SOURCE  -O6 
-march=i686  -Wno-missing-prototypes -Wno-missing-declarations 
-DCRYPTO   -c -o chan_capi.o chan_capi.c

In file included from chan_capi.c:82:
chan_capi.h:41:2: warning: #warning If you dont update your libcapi20, 
some fax features are not available
chan_capi.c:146: warning: type defaults to `int' in declaration of 
`STANDARD_LOCAL_USER'

chan_capi.c:146: warning: data definition has no type or storage class
chan_capi.c:147: warning: type defaults to `int' in declaration of 
`LOCAL_USER_DECL'

chan_capi.c:147: warning: data definition has no type or storage class
chan_capi.c: In function `capi_new':
chan_capi.c:2069: error: too few arguments to function `ast_channel_alloc'
chan_capi.c:2083: error: structure has no member named `type'
chan_capi.c: In function `pbx_capicommand_exec':
chan_capi.c:4613: warning: implicit declaration of function `LOCAL_USER_ADD'
chan_capi.c:4628: warning: implicit declaration of function 
`LOCAL_USER_REMOVE'

chan_capi.c: At top level:
chan_capi.c:5275: error: unknown field `send_digit' specified in initializer
chan_capi.c:5275: warning: initialization from incompatible pointer type
make: *** [chan_capi.o] Error 1

//--

Since the configuration method is a bit too much for me, here's part of 
chan_capi Makefile. I think I've been blind as I haven't found the 
documentation for WHAT needs to go WHERE in this Makefile...


.PHONY: openpbx

INSTALL_PREFIX=/usr/lib/asterisk

ASTERISK_HEADER_DIR=/usr/src/asterisk-1.4.0/include

MODULES_DIR=/usr/lib/asterisk/modules

CONFIG_DIR=/etc/asterisk


//--

If anyone has any idea what I'm doing wrong, please help me,
Thanks,
Cosmin Prund
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Re: [asterisk-users] Failing to compile chan_capi

[Cosmin Prund]

This gets fixed using the chan_capi.HEAD, not chan_capi.0.7.1; chan_capi 
compiled right out of the tar.gz, no changes required (the defaults in 
the Makefile are ok)


Cosmin Prund wrote:
I've got a brand new Eicon Diva Server BRI card and I want to 
configure it with Asterisk. I managed to get asterisk and zaptel to 
compile and install, I've compiled and installed the drivers for the 
Diva card and now I need to compile and install the chan_driver for 
chan_capi. Unfortunately this fails miserably. I get the following 
messages:


I'm using: Kernel 2.6.16.37.4, zaptel-1.4.0 and asterisk-1.4.0., 
chan_capi 0.7.1


//-- 



[EMAIL PROTECTED] chan_capi-0.7.1]# make
./create_config.sh /usr/src/asterisk-1.4.0/include
Checking Asterisk version... 1.4.0
* found stringfield in ast_channel
* found 'indicate' with data
config.h complete.
gcc -pipe -fPIC -Wall -Wmissing-prototypes -Wmissing-declarations -g  
-I/usr/src/asterisk-1.4.0/include -D_REENTRANT -D_GNU_SOURCE  -O6 
-march=i686  -Wno-missing-prototypes -Wno-missing-declarations 
-DCRYPTO   -c -o chan_capi.o chan_capi.c

In file included from chan_capi.c:82:
chan_capi.h:41:2: warning: #warning If you dont update your libcapi20, 
some fax features are not available
chan_capi.c:146: warning: type defaults to `int' in declaration of 
`STANDARD_LOCAL_USER'

chan_capi.c:146: warning: data definition has no type or storage class
chan_capi.c:147: warning: type defaults to `int' in declaration of 
`LOCAL_USER_DECL'

chan_capi.c:147: warning: data definition has no type or storage class
chan_capi.c: In function `capi_new':
chan_capi.c:2069: error: too few arguments to function 
`ast_channel_alloc'

chan_capi.c:2083: error: structure has no member named `type'
chan_capi.c: In function `pbx_capicommand_exec':
chan_capi.c:4613: warning: implicit declaration of function 
`LOCAL_USER_ADD'
chan_capi.c:4628: warning: implicit declaration of function 
`LOCAL_USER_REMOVE'

chan_capi.c: At top level:
chan_capi.c:5275: error: unknown field `send_digit' specified in 
initializer

chan_capi.c:5275: warning: initialization from incompatible pointer type
make: *** [chan_capi.o] Error 1

//-- 



Since the configuration method is a bit too much for me, here's part 
of chan_capi Makefile. I think I've been blind as I haven't found 
the documentation for WHAT needs to go WHERE in this Makefile...


.PHONY: openpbx

INSTALL_PREFIX=/usr/lib/asterisk

ASTERISK_HEADER_DIR=/usr/src/asterisk-1.4.0/include

MODULES_DIR=/usr/lib/asterisk/modules

CONFIG_DIR=/etc/asterisk


//-- 



If anyone has any idea what I'm doing wrong, please help me,
Thanks,
Cosmin Prund
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[asterisk-users] About BRI / ISDN hardware. What to buy?

[Cosmin Prund]

Hello everyone.

I need a BRI ISDN card that works in Romania. I already have one of the 
Cologne HFC-S PCI cards and it doesn't work right, it's junk. I get 
wy too much echo using it. I'm now shopping for a better card. Can 
anyone recommend me a card that fits the following:


(a) Costs less then $1000 / 750 euro
(b) Has one or (preferably) two ISDN S0 interfaces.
(c) Easy to set up.
(d) Drivers offer proper echo-canceling OR has an hardware echo 
canceler. I might increase the $1000 a bit if I can get good hardware 
echo canceler...


Thanks,
Cosmin Prund
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Re: [asterisk-users] About BRI / ISDN hardware. What to buy?

[Cosmin Prund]

I do not care about fax. I just want a good VOICE card.

Can someone please give a price quote for this card, give or take 10%? I 
just spent 5 minutes filling in a really long form on a shopping web 
site to get a price quote, only to find my account needs to be manually 
activated before I can see the price! That's *STUPID*. If I have a 
choice, I'll buy it from somewhere else...


Jens Vagelpohl wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1


On 18 Jan 2007, at 18:31, Patrick wrote:

I think http://www.melware.de carries the Eicon Server ISDN cards which
have hardware echo cancellation. They are also the author of the
chan_capi driver for Asterisk. I use the Eicon Server BRI cards with
Asterisk myself and they work very well.


I concur, I have a Eicon DIVA single port BRI card and it works very 
well.


Cosmin, if you want to use it for Fax traffic as well make sure you do 
*not* get a V-BRI card. Those will not do Fax.


jens



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Version: GnuPG v1.4.5 (Darwin)

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8PZj2V5wU201Eu/+U/W1ihM=
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Re: [asterisk-users] About BRI / ISDN hardware. What to buy?

[Cosmin Prund]

How about the Digium Wildcard B410P card? It seems to be Digium, it 
has hardware echo cancel and I can buy this in Romania. Is this card any 
good?


Cosmin Prund wrote:

Hello everyone.

I need a BRI ISDN card that works in Romania. I already have one of 
the Cologne HFC-S PCI cards and it doesn't work right, it's junk. I 
get wy too much echo using it. I'm now shopping for a better 
card. Can anyone recommend me a card that fits the following:


(a) Costs less then $1000 / 750 euro
(b) Has one or (preferably) two ISDN S0 interfaces.
(c) Easy to set up.
(d) Drivers offer proper echo-canceling OR has an hardware echo 
canceler. I might increase the $1000 a bit if I can get good hardware 
echo canceler...


Thanks,
Cosmin Prund
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Re: [asterisk-users] About BRI / ISDN hardware. What to buy?

[Cosmin Prund]

I finally found a price tag for the darn thing, at around 500 euros I 
can handle it.
Qustion: Do they behave properly if I've got an other Digium TDM400 card 
in the system? How about installing two cards in the same server?
At the moment I've only got 1 ISDN line plus a few analog lines going 
into the TDM but in the very near future we might want to get a second ISDN.


Alberto Pastore wrote:

Jens Vagelpohl ha scritto:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1


On 18 Jan 2007, at 18:31, Patrick wrote:

I think http://www.melware.de carries the Eicon Server ISDN cards which
have hardware echo cancellation. They are also the author of the
chan_capi driver for Asterisk. I use the Eicon Server BRI cards with
Asterisk myself and they work very well.


I concur, I have a Eicon DIVA single port BRI card and it works very 
well.


Cosmin, if you want to use it for Fax traffic as well make sure you 
do *not* get a V-BRI card. Those will not do Fax.


jens



Tried almost all cards (Junghanns, Sangoma, Beronet, some hfc-based oem
cards, Eicon Diva Server).

Eicon is expensive but is *REALLY* worth it.
The other cards are just a waste of money (even if little money).

If you want a reliable PBX (who doesn't want it?),
Diva Server cards are the definitive choice.

The best card ever.
Zero echo problems, superb hardware echo cancellation.
Top reliability.
Excellent FAX support with Hylafax (only cards with builtin DSPs,
that is, NOT the V-series, as pointed out by Jens).

Easy driver installation and powerful utilities/configuration tools.


I tested BRI-2M, 4BRI-8M, PRI-30M on several installations,
even older 1.0 version cards (PCI 5v only) just work great.

I use diva server drivers  software source rpm from Eicon,
chan_capi from www.melware.org (0.7.1) on asterisk 1.2.14
(kernel 2.6.17.3). We've deployed more than 40 PBX (from 1 bri
to 8 bri) without a flaw.

I'm only a little bit annoyed about not being able to take
advantage of the onboard DSPs to perform audio transcoding,
because
of the lack of a suitable asterisk driver
(the cards themselves support hardware gsm/g726 codecs,
for instance).

Alberto.

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[asterisk-users] Install asterisk-bristuff for Debian Linux

[Cosmin Prund]

Following a discussion on this list about a week ago I downloaded and 
installed Debian Linux. Now I want to install asterisk-bristuff.

How do I do that?

Better yet, what do I put in /etc/apt/sources.list so I can do 
apt-get install asterisk-bristuff


--
Thanks for your help,
Cosmin Prund
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Re: [asterisk-users] Install asterisk-bristuff for Debian Linux

[Cosmin Prund]

Thanks! It works! (at first)

I installed my deb from the given repository and I think it all went 
find. Asterisk starts up and I can get to the console. But... where are 
the drivers? updatedb / locate sees no zaptel drivers, and I've got none 
of the zapp tools on the system. Is that a separate download/install? If 
so, what's the name of the package I need to install?


Thanks

Filip Drągowski wrote:

Google is Your friend
http://peen.net/2006/04/15/asterisk-1271-and-zaptel-125-for-debian-sarge/

Following a discussion on this list about a week ago I downloaded and 
installed Debian Linux. Now I want to install asterisk-bristuff.

How do I do that?

Better yet, what do I put in /etc/apt/sources.list so I can do 
apt-get install asterisk-bristuff


--
Thanks for your help,
Cosmin Prund


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[asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?

[Cosmin Prund]

I've been trying to install bristuff on my system for a really long time.
This is what I've done so far:

I started with a [EMAIL PROTECTED] installation. I tried downloading and 
compiling bristuff release - it didn't work. It was a long time ago, I 
don't remamber what the problem was.
I tried compiling the latest bristuff (whatever latest was about 1-2 
months ago). It failed to compile.


I download the full CentOS 4.3 and tried compiling both bristuff release 
(0.2.0-RC8) and bristuff latest. Again, whatever latest was about 2 
weeks ago.


Next I found something about bristuff being known to work on kernel 2.4; 
Since CentOS 4.3 has kernel 2.6 I downloaded CentOS 3 and tried 
compiling both bristuff release (0.2.0-RC8) and the current release of 
today (19 july 2006). I wasn't able to compile ither one of them.


Now, after downloading 2 DVD Linux images (CentOS 4.3 and CentOS 3) and 
two CD Images ([EMAIL PROTECTED] and Trixbox) and reinstalling everything a 
few times, I think it's time to ask for help:


Would someone be so kind and tell my how they installed Bristuff from A 
to Z? (that is, what version of Linux so I can download the same 
version, what updates, what version of bristuff). I'm hoping for a quick 
answer like: Install LinuxVariant 10.20, install all updates using 
LinuxVariantUpdateProgram, download bristuff version X.Y.Z, call 
install.sh and be done with it.


Thanks,
Cosmin Prund
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Re: [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?

[Cosmin Prund]

Thanks for your input. I'm good at following instructions (if I can find 
instructions) so I'll give anything a try!


I'm downloading bristuff from http://junghanns.net/downloads/ and the 
tar's I'm getting from there contain some kind of bootstraping for the 
installation. The install.sh file simply calls download.sh and then 
compile.sh.


The download.sh script downloads a specific version of asterisk (and 
everything else required) so I doubt it gets it wrong. It then patches 
the thing all by itself, using exactly the instructions you gave. It 
fails when it tries compiling stuff. I installed CentOS 3 using 
everything as an install option so I think I've got everything.


If you ever got this working, would you be so kind to tell what version 
of Linux you used and what version of bristuff? I prefer 
CentOS/Fedora/RHL instalations as that's what I've always used and 
that's what I know, but I'm willing to use anything as long as it gets 
the work done.


Thanks again,
Cosmin Prund

Filip Drągowski wrote:

Bristuff 0.2.0-RC8 if for Asterisk 1.0.10
Bristuff 0.3.0-PRE-1r if for Asterisk 1.2.9.1, libpri 1.2.3 and zaptel 
1.2.6

download proper versions
(for Asterisk 1.2.9.1)
look at install.sh in bristuff directory
do as it's written there:
cd zaptel
patch -p1  ../patches/zaptel.patch
cd ..

cd libpri-1.2.3
patch -p1  ../patches/libpri.patch
cd ..

cd asterisk-1.2.9.1
patch -p1  ../patches/asterisk.patch
cd ..

then try to install Trixbox.

[EMAIL PROTECTED] as i kno comes with whole OS and asterisk installation 
is automated and ther is no time for applying bristuff patches.

It looks that You have to manually install OS and asterisk then trixbox

-Hope that help You a little.
I've been trying to install bristuff on my system for a really long 
time.

This is what I've done so far:

I started with a [EMAIL PROTECTED] installation. I tried downloading and 
compiling bristuff release - it didn't work. It was a long time ago, 
I don't remamber what the problem was.
I tried compiling the latest bristuff (whatever latest was about 
1-2 months ago). It failed to compile.


I download the full CentOS 4.3 and tried compiling both bristuff 
release (0.2.0-RC8) and bristuff latest. Again, whatever latest was 
about 2 weeks ago.


Next I found something about bristuff being known to work on kernel 
2.4; Since CentOS 4.3 has kernel 2.6 I downloaded CentOS 3 and tried 
compiling both bristuff release (0.2.0-RC8) and the current release 
of today (19 july 2006). I wasn't able to compile ither one of them.


Now, after downloading 2 DVD Linux images (CentOS 4.3 and CentOS 3) 
and two CD Images ([EMAIL PROTECTED] and Trixbox) and reinstalling 
everything a few times, I think it's time to ask for help:


Would someone be so kind and tell my how they installed Bristuff from 
A to Z? (that is, what version of Linux so I can download the same 
version, what updates, what version of bristuff). I'm hoping for a 
quick answer like: Install LinuxVariant 10.20, install all updates 
using LinuxVariantUpdateProgram, download bristuff version X.Y.Z, 
call install.sh and be done with it.


Thanks,
Cosmin Prund


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Re: [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?

[Cosmin Prund]

I don't use Debian but I'm going to give Debian a try. I'm downloading
the Debian distribution right now. Unfortunatelly it will be about a
week till I'll download the whole 8Gb (2xDVD iso).

Tzafrir Cohen wrote:

  On Wed, Jul 19, 2006 at 03:17:15PM +0200, Filip Drągowski wrote:
  
  
First question: Do You have kernel sources ?
this is required for #make-ing zaptel

i have installed asterisk-1.2.4 with bristuff-0.3.0-PRE-1k, libpri-1.2.2 
and zaptel-1.2.3

OS was Debian 3.1 with kernel 2.6.16.2, i compiled new kernel myself so 
there was kernel sources in system.
I didn't use bristuff autamated install.
wget-ed asterisk, libpri, zaptel and patched them.
there is recomended to use make linux26 when making zaptel on 2.6. 
kernel. bristuff compile.sh don't have linux26 option

  
  
If you use Debian, you'd probably be better off with the bristuff
asterisk debs. They get automatically built for Sarge as well...

  




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Re: [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?

[Cosmin Prund]

I'm using a HFC-S ISDN card and a TDM400P zaptel card with 3xFXO and 
1xFXS modules.
How did you finally manage to compile bristuff on Centos 4? I'm 
downloading Debian right now but the 2 DVD images will take about a week 
to download so I'm willing to try anything else in the time. I've got 
both the binary and the source DVD's for CentOS 4 if that makes any 
difference...


Kai Ober wrote:

Filip Drągowski schrieb:

First question: Do You have kernel sources ?
this is required for #make-ing zaptel

i have installed asterisk-1.2.4 with bristuff-0.3.0-PRE-1k, 
libpri-1.2.2 and zaptel-1.2.3


OS was Debian 3.1 with kernel 2.6.16.2, i compiled new kernel myself 
so there was kernel sources in system.

I didn't use bristuff autamated install.
wget-ed asterisk, libpri, zaptel and patched them.
there is recomended to use make linux26 when making zaptel on 2.6. 
kernel. bristuff compile.sh don't have linux26 option


that linux26 stuff is as far as i know only important to ztdumm.ko, a 
kernel module which is needed, if you have no Zaptel Cards in your PC

and want to use MeetMe Conferencing system.

you dont need to tell zaptel wheter you have a 2.6 or 2.4 Kernel, the 
Makfile discovers this himself.


so, no need to worry about 2.4 or 2.6 stuff.

Getting kernel sources was a torture for me on Cent-OS 4.
maybe somebody can explain how to get them the right way!!!
and apply the patches and that.

Which Cards do you wanna use in your asterisk
(especiallly which ISDN cards, if any)

can you post the errormessage of the bristall install script?


regards Kai





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Re: [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?

[Cosmin Prund]

Tzafrir Cohen wrote:

  On Wed, Jul 19, 2006 at 09:30:30PM +0300, Cosmin Prund wrote:
  
  
I don't use Debian but I'm going to give Debian a try. I'm downloading 
the Debian distribution right now. Unfortunatelly it will be about a 
week till I'll download the whole 8Gb (2xDVD iso).

  
  
Unless you have a bad internet connection, just grab the first ISO , or
even the netinst ISO image. If you install it in a network with a DHCP
server and internet access, that' basically all you need. The packages
you'll actually need will be downloaded at install time.
  

Got bad internet connection :-( - only 192 kbit


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Re: [Asterisk-Users] Getting the Server IP

[Cosmin Prund]

Your best bet is to write it by hand. If you don't want that you may use 
an AGI script to interrogate the operating system, selecting the one IP 
address you like the most. Please take into account system's with more 
then one net card. Please also note you may have more then one IP 
address per net card :-)


Steven Ringwald wrote:

Hello all!

Can anyone think of an *easy* way to get the IP number of the server 
running asterisk from within the dialplan?


Thank you in advance!
Steve

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Re: [Asterisk-Users] Are my expectations too high?

[Cosmin Prund]

Thanks for the tips on changing things myself and thanks for the tip on 
the Digium changes. I guess I'll need to make the changes myself as I 
need to use Bristuff and I don't think I'm getting the 
latest-and-gratest :-)


Andrew Kohlsmith wrote:

On Tuesday 23 May 2006 16:28, Andrew Kohlsmith wrote:
  

On Tuesday 23 May 2006 14:54, Cosmin Prund wrote:


Is there a known hack or patch to blow Asterisk's echo canceler up to
128ms? Or at least 64 ms?
  

It's real easy... at least for us code tinkerers.



It's even easier now, as Digium has just committed these changes to svn trunk 
(within the last hour).  It looks like the defaults will remain the same, but 
you will be able to specify up to 1024 taps (128ms) with just a config file 
change.


-A.
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Re: [Asterisk-Users] Are my expectations too high?

[Cosmin Prund]

Andrew Kohlsmith wrote:
Regarding echo tail: Asterisk's default software echo cancellation is only 
32ms at maximum.  You can blow this number up to 128ms (where the hardware 
echo cancellers sit at), but it costs more in CPU time and memory, and I 
think that Digium left it at 32ms as a compromise.  I'm not sure.
  
Is there a known hack or patch to blow Asterisk's echo canceler up to 
128ms? Or at least 64 ms?


I do understand this would not work for medium/large Asterisk servers 
but my small-home-office server can't possibly go over two simultaneous 
echo-canceled calls because that's the number of POTS lines I've got! My 
CPU is almost IDLE with two simultaneous SIP-to-POTS calls - I see that 
as waisted resources :-)

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Re: [Asterisk-Users] FXS Caller ID revisted

[Cosmin Prund]

Are you doing something funny with the CID on it's way to the phone? 
I've got a somewhat similar problem with an Aastra IP phone (yes, I did 
say IP): it would NOT ring if the caller id started with an #. Maybe 
your Aastra PSTN phone got some of the same (buggy?) handling of CID's?


Dan Elder wrote:

Hi All, posted last week but didn't get any responses. I'm trying to figure
out why some of our analog phones aren't showing CID when hooked up to
asterisk. To recap, I have an Aastra Powertouch 350, which shows caller ID
fine when connected to the PSTN, but when hooked up to asterisk, CID does
not show. I've hooked up another phone to the same * port that the Aastra
phone is on,  it DOES show CID, so I'm assuming my settings  such are at
least partially correct, can anyone point me to some options or areas I can
look to troubleshoot this issue? Been pulling my hair out on this for days 
just can't seem to get it sorted.

I'm using asterisk 1.2.0 with a Carrier Access ABII channel bank. When
another CID capable phone is hooke up to the same port, CID works fine, the
Aastra phone is however unable to read the incoming CID from * apparently.

Any pointers would be greatly appreciated, I've searched the Wiki  the CID
faq's to no avail.

Thanks in advance

Dan

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Re: [Asterisk-Users] hardware help ?

[Cosmin Prund]

Take a look on the Digium site for the requirements of the card. Most 
likely the old computer can't handle the card.


M.Hockings wrote:
I just bought an TDM400P card with one FXS port and a X101P FXO card 
to try and put together the beginnings of a PBX here.  The computer 
they are going in should arrive in a week.  So I thought I would start 
learning how to config the card in an IBM Aptiva we have here, an old 
400Mhz P2 box.  However when I put the TDM400P in the box the machine 
won't POST.  Does this mean that the card is bad or that the machine 
can't handle this new PCI hardware?


If it matters the cards were purchased online at voipdepot.ca just 
this week and arrived this morning.


Mike

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Re: [Asterisk-Users] Experience with IBM X346 machines and Sangoma

[Cosmin Prund]

I wanted to see where those periodical spikes are coming from so I 
started shutting things down. The first thing to go was Asterisk. Next 
went many different things like mysql and ntpd. Finally I killed zaptel 
(/etc/init.d/zaptel stop) - and the spiking stoped!


Next I rebooted and I've done /etc/init.d/zaptel stop straight away. 
The spiking stoped again. I've done /etc/init.d/zaptel start and 
spiking started again!


Is there something funny happening with my zaptel?
Wolfgang Zweimueller, can you give this a try too? Does your spiking 
stop when you stop zaptel?

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Re: [Asterisk-Users] Default dialplan??

[Cosmin Prund]

I'll give this one a try, but don't trust me, test it yourself :-)

Of course Asterisk can do it! All you need to do is set up a rule for 
matching ALL extensions in the PBX in it's own separate context and 
include that context into your normal context. In the following 
example, asterisk will try matching all extensions in context Normal 
(all extensions defined on *) and, if no match was found, start 
searching the context secondary_pbx. In my sample this secondary 
context will match any 3-digit number and send it to the other PBX. 
Should work...


[Normal]
include = secondary_pbx
exten = 101,1,Dial(sip/101)

[secondary_pbx]
exten = _XXX,Dial(Zap/g1)

Aaron Paxson wrote:

Hey all!
 
I've got my Asterisk box tied into my PBX.  Currently, if a call comes 
into my PBX, and can't find the extension, it forwards it through my 
Asterisk trunk to Asterisk.
 
This works great!
 
Is there a special dialplan function (or common usage pattern) that 
can do the same thing in Asterisk?  i.e. If it can't find the 
extension, send it out Zap/g1?
 
My dialplan works with patterns, but patterns isn't what I need here.  
Is anyone doing anything like this?
 
Thanks!

~~Aaron


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Re: [Asterisk-Users] New To Asterisk - Advice needed

[Cosmin Prund]

I'm fairly new to Asterisk myself and I also started with AAH. 
Unfortunately I had to remove all configuration files generated by 
FreePBX (the GUI of AAH) and started over using http://voip-info.org as 
my guide. Configuration files generated with FreePBX make use of 
advanced functionality available in Asterisk and that in turn makes it 
hard (impossible?) to read for a newby. If you've got some experience 
with Linux and it's kind of configuration files you might be better of 
without AAH. On the other hand I'm in the process of re-installing my 
Asterisk on a fresh Centos 4.3 installation so I can't comment on how 
difficult it is for a newby to install everything from sources. Hope 
I'll be able to manage it :)


Mark Adams wrote:


Hi People,

I’m writing to get some advice on where to start when learning 
asterisk? I was going to begin learning with AAH but I wanted to find 
out if there is a certain build to avoid or if there is a Gui/front 
end that is better then another. I have been working with dialogic 
cards for the past 5 years and with auto dialers but I want to get 
into providing voip service, support and eventually help people save 
money with their phone systems. At the moment it is strictly for 
education but I really get a kick out of voip and telephone functions 
in general.


Thanks in advance

- Mark Adams




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Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?

[Cosmin Prund]

Ralf Schlatterbeck wrote:

On Fri, May 12, 2006 at 02:25:44PM +0300, Cosmin Prund wrote:
  
Unfortunately the latest misdn-mqueue does not compile on my system, it 
issues all sorts of blah-blah that I'm interpreting in only one way: 
there's a problem with the parameters the Makefile passes to the 
compiler (the .h files where the error manifests itself are part of 
Asterisk and compile fine when compiled with Asterisk itself).


The mqueue branch was merged to head some time ago. Maybe you want to
try the HEAD of misdn. mqueue is dead.

Ralf
  

Thanks for your input.
Where do I get HEAD from?
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[Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?

[Cosmin Prund]

Hello everyone.

I've got a HFC ISDN card that I'm using with chan_misdn and it basically 
behaves like crap. Echo is waaay worst then echo I get TDM400 card, 
sound is choppy (there other side is allays complaining about sound 
interruptions) and to top it all it detects fake DTMF's all the time.


Is this a chan_misdn problem or is it a card problem? I really need to 
get this fix and I need to know the way to go. I don't want to throw 
money at a better card if the card is not the issue but if that's the 
only solution, I'll need to order the card ASAP!

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Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?

[Cosmin Prund]

Woodoo People .pGa! wrote:
I've got a HFC ISDN card that I'm using with chan_misdn and it basically 
behaves like crap. Echo is waaay worst then echo I get TDM400 card, 
sound is choppy (there other side is allays complaining about sound 
interruptions) and to top it all it detects fake DTMF's all the time.


Is this a chan_misdn problem or is it a card problem? I really need to 
get this fix and I need to know the way to go. I don't want to throw 
money at a better card if the card is not the issue but if that's the 
only solution, I'll need to order the card ASAP!



i'm using 1port (billion bipac), quad and octoBRI cards from beronet.
all of them working nice, beronet recommend to use kernel 2.6.12+ and
asterisk 1.2.x and also newest misdn-mqueue from www.beronet.com
  
Seeing the names on this list I realize I've tried lots and lots of 
different things. I'm running kernel 2.6.15.11 so I'm above 12. 
Unfortunately the latest misdn-mqueue does not compile on my system, it 
issues all sorts of blah-blah that I'm interpreting in only one way: 
there's a problem with the parameters the Makefile passes to the 
compiler (the .h files where the error manifests itself are part of 
Asterisk and compile fine when compiled with Asterisk itself).


Those are the errors I get:

./create_config.sh /usr/include
Checking Asterisk version...
* found 'struct ast_channel_tech'
* found 'ast_bridged_channel'
* found 'ast_bridge_result'
* found bridge with timeoutms
* ast_dsp_process() without 'needlock'
* found 'struct ast_callerid'
* found 'struct timeval delivery'
* found 'transfercapability'
* found 'ast_config_load'
* found 'AST_CONTROL_HOLD'
* found 'devicestate.h'
* found 'strings.h'
* no 'type' in ast_channel
* found stringfield in ast_channel
config.h complete.
gcc -pipe -fPIC -Wall -Wmissing-prototypes -Wmissing-declarations -g  
-I/usr/include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686  
-Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO   -c -o 
chan_capi.o chan_capi.c

In file included from /usr/include/asterisk/utils.h:36,
   from /usr/include/asterisk/cdr.h:48,
   from /usr/include/asterisk/channel.h:113,
   from chan_capi.c:23:
/usr/include/asterisk/strings.h:264: error: syntax error before 
__extension__

/usr/include/asterisk/strings.h:264: error: syntax error before ';' token
/usr/include/asterisk/strings.h:264: error: `__len' undeclared here (not 
in a function)
/usr/include/asterisk/strings.h:264: error: initializer element is not 
constant

/usr/include/asterisk/strings.h:264: error: syntax error before if
/usr/include/asterisk/strings.h:264: error: redefinition of '__retval'
/usr/include/asterisk/strings.h:264: error: previous definition of 
'__retval' was here

/usr/include/asterisk/strings.h:264: error: syntax error before const
/usr/include/asterisk/strings.h:264: error: syntax error before '}' token
/usr/include/asterisk/strings.h:280: error: conflicting types for 'strtoq'



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Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vsexpensive card?

[Cosmin Prund]

Chris Bagnall wrote:
I've got a HFC ISDN card that I'm using with chan_misdn and 
it basically behaves like crap. Echo is waaay worst then echo 
I get TDM400 card, sound is choppy (there other side is 
allays complaining about sound

interruptions) and to top it all it detects fake DTMF's all the time.
Is this a chan_misdn problem or is it a card problem? 



We have a number of sites running from 1-3 HFC-based cards in a machine, and
none of them have any significant echo at all. All ours are running with
zaphfc (part of the bristuff package). Might be worth giving that a try.

Regards,

Chris
  
Thanks for the info, I'm compiling bristuff 0.3 right now, hope it 
works. I'll need to wait for the compile to finish to see how it works 
with my kernel (as my kernel has been patched for mISDN and I do not 
know how that plays with bristuff).

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[Asterisk-Users] mISDN trouble with a HFC Cologne card, Asterisk Asterisk 1.2.4 on Linux 2.6.16.11 - incoming DTMF detection

[Cosmin Prund]

Hello everyone. I've got this really annoying HFC Cologne card (or 
however I should call it - a single channel ISDN card based on the HFC 
chipset).
It wrongfully detects lots and lots and lots of incoming DTMFs, to the 
point the card is not usable.


Here's a sample out of CLI:

P[ 1] I IND :DTMF_TONE oad:206361 dad:520101
P[ 1]  -- mode:TE cause:16 ocause:16 rad: cad:
P[ 1]  -- facility:FAC_NONE out_facility:FAC_NONE
P[ 1]  -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0
P[ 1]  -- screen:0 -- pres:0
P[ 1]  -- channel:1 caps:Speech pi:2 keypad:
P[ 1]  -- urate:0 rate:16 mode:0 user1:0
P[ 1]  -- pid:1 addr:50010102 l3id:30001
P[ 1]  -- b_stid:10010100 layer_id:50010180
P[ 1]  -- bc_state:BCHAN_ACTIVATED
P[ 1]  -- DTMF:*

What's this all about? Is there anything I can do about it?
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Re: [Asterisk-Users] asterisk management interface

[Cosmin Prund]

You can't possibly read conf files and present them to the user in a 
web-based form because of the complexity of the data in those file. Take 
the extensions configuration file for an example (as this is probably 
the most complex one). Settings in this file might specify a simple 
mapping between extensions and physical phones but it might as well 
include other things. Ex: dialing 500 on my PBX will reboot the server! 
Dialing 199 will ring two phones. You might also have IVR's in a an 
extensions.conf and very complex call-handling logic based on almost 
anything you can think of. There's plain simply no way of mapping that 
wealth of info into some structured form into a web page!


What I think is a better idea is to store ALL your info into a database 
and re-generate the configuration files whenever that data changes. You 
can get a very customizable system in place this way but, unless your 
needs are somewhat different you might be better off using one of the 
pre-built systems.


moona ather wrote:

Hi,
I have to make a web-based management interface of configuring 
asterisk i wanted to know if it is as simple as reading the .conf 
files and searching for the required section in the file and adding 
users etc. or there are other steps involved too?? As I have seen many 
such built codes on this site and found lots of code... kindly tell me 
how complex it is and how many other steps are involved in making this 
interface as i am new in this.

Emmo.

_
Don't just search. Find. Check out the new MSN Search! 
http://search.msn.click-url.com/go/onm00200636ave/direct/01/


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Re: [Asterisk-Users] How do I monitor a Zap channel ...

[Cosmin Prund]

I'm using ChanSpy. Set up an extension with ChanSpy, dial the given 
extension and don't hang up (put it on speakerphone). When there's no 
one on the given zap channel you'll here silence. As soon as someone's 
on the channel you'll be listening to them.


If you don't like ChanSpy there's ZapBarge. And if by monitor you mean 
record there's Monitor!


Anthony Azzopardi wrote:
How do I monitor a Zap channel as soon as the telephone is off the 
hook, till it is on the hook again?

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[Asterisk-Users] Gigabit Ethernet with multiple VLAN's or Fast Ehternet and with two separate cards?

[Cosmin Prund]

Hello everyone.

What's better for Asterisk: have 2 distinct 100Mb network cards in the 
system, one on the internet and one on the local net OR have one 
1000Mb network card with 2 separate VLAN's set up? It's a difficult 
decision because 2 cards are using 2 IRQ's etc but a single 1000Mb card 
might generate more PCI interrupts and get me into different kinds of 
problems.


I'm leaning towards the gigabit card because my switch supports it and 
I've seen no references to problems caused by a gigabit card + tdm400 etc.

Any ideas?

Thanks!
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RE: [Asterisk-Users] Help on chan_misdn and MSN's

[Cosmin Prund]

I've got my MSN's going, I'll just share how I did it below:

My initial assumption was wrong. I'm supposed to have one section per ISDN
channel listing all the MSN's chan_misdn is responsible for. When one of
those MSN's is detected chan_misdn is supposed to jump into the dialplan in
the specified context at the extension specified in the MSN.

What I did was fairly simple. First of all I had to set immediate=yes.
Unless I had that option chan_misdn would not pick up incoming ISDN calls.

With that option set to yes chan_misdn did exactly what the documentation
sad: It jumped in the specified context at the s extension. So my dialplan
would receive no info on the called MSN.

Next I entered the directory where I had the sources for chan_misdn and
griped for Starting Ast ctx. It only appears in one file. Three lines
lower in the source file is a line that changes the extension to s. I
simply commented out that line, rebuilt chan_misdn and voila: I've got my
MSN's in the dialplan!

Finally I'm not sure I found a small compatibility problem between
chan_misdn and the Romanian implementation of ISDN or I simply solved a
configuration problem with a huge hammer but I'm happy it works! 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Cosmin Prund
 Sent: Tuesday, April 25, 2006 10:41 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Help on chan_misdn and MSN's
 
 Quick question:
 Is there a way to distinguish between calling MSN's when using chan_misdn?
 
 More info:
 I've got my ISDN2 (EuroISDN) up and running here in Romania with 1 base
 number plus 5 MSN's. Now I want to my * to do different things when
 receiving a call on from different MSN's (like forwarding the call to my
 FAX
 machine or forwarding the call to my mobile).
 
 The obvious way of doing this would be to set up different sections in
 the
 misdn.conf file for the same port (I only have one port), using different
 settings for the msns. Unfortunately it seems that the channel driver will
 only remember the last section it sees for a given channel so I can only
 use
 * as the msn - and that defeats the purpose.
 
 If any other info is required I'll happily provide it. I'm not including
 any
 other info at the moment because I'm unable to filter the list myself
 and
 the list of things I've been doing today is very long (starts with
 downloading kernel 2.6.16.11 off kernel.org, patching for mISDN,
 downloading
 chan_misdn, compiling everything... waaay too long list, most of it
 irrelevant)
 
 
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[Asterisk-Users] Help on chan_misdn and MSN's

[Cosmin Prund]

Quick question:
Is there a way to distinguish between calling MSN's when using chan_misdn?

More info:
I've got my ISDN2 (EuroISDN) up and running here in Romania with 1 base
number plus 5 MSN's. Now I want to my * to do different things when
receiving a call on from different MSN's (like forwarding the call to my FAX
machine or forwarding the call to my mobile).

The obvious way of doing this would be to set up different sections in the
misdn.conf file for the same port (I only have one port), using different
settings for the msns. Unfortunately it seems that the channel driver will
only remember the last section it sees for a given channel so I can only use
* as the msn - and that defeats the purpose.

If any other info is required I'll happily provide it. I'm not including any
other info at the moment because I'm unable to filter the list myself and
the list of things I've been doing today is very long (starts with
downloading kernel 2.6.16.11 off kernel.org, patching for mISDN, downloading
chan_misdn, compiling everything... waaay too long list, most of it
irrelevant)


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[Asterisk-Users] Callback auto dialing

[Cosmin Prund]

Hello everyone.

This is an other question from a relatively newbie. 

I'd like to provide auto callback ability for my *. From my mobile I want to
be able to call a number on the * and have it call me back on my mobile. I
know how to generate a .call file from a script and I know how to call a
script from the dialplan (in order to get the .call file generated). I also
found the scripts on www.voip-info.org on callback voicemail but what I
want is not voicemail. I just want to talk to the * and use it's much lower
rates!

What I do not know is what to write in that call file so I'll get an IVR
when I answer the phone, not Voicemail or an other channel. It seems that
call files are designed to connect one channel to an other channel or one
channel to an application. But I don't want to connect to an application
like Voicemail, I want the system to behave as if I called the other way
around and ended up into an arbitrary context.

Thanks for any help,
Cosmin Prund,
Romania


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RE: [Asterisk-Users] call transfer to external phone number

[Cosmin Prund]

 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of C F
 Sent: Monday, April 03, 2006 3:49 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] call transfer to external phone number
 
 Yes, as long as the context that the phone transfering has an exten
 declared for that number.
 

Does Asterisk make any distinction between an internal number and an
external number? I'm inclined to think it might be some kind of timeout
issue. And I've got the proof:

From my Aastra 9xxx (don't know the number and I'm to lasy to go look) I can
transfer a call to any extension, including the lng extension required
for dialing an external number (ie: #0X). Unfortunatelly that's the
ONLY phone I can do that from! I can't do it from XLite softphone and I
can't do it from analog phones connected to a Linksys PAP2.

For the phones that are unable to transfer to external numbers I've got
alias extensions defined (basic, 3 digit extensions).

 On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote:
  Hi!
  Is it possible to transfer a call to an external phone instead of
  transferring the call to internal phone?
  (I'm sorry for my bad english, I hope you understand)
  When, during a call, I digit #123, the call is transferred to internal
  extension 123,
  but if I digit #external_phone_number, it tells me that it's
 impossible.
  Any idea?
 
  Thanks a lot!
 
  Giuseppe
 
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[Asterisk-Users] Coice recognition IVR?

[Cosmin Prund]

Hello everyone.

Is it possible to do some very basic voice recognition from within
Asterisk's dialplan? What I'm aiming at is the ability to speak the digits I
want to dial from my mobile phone. Dialing digits on my mobile phone while
driving is not all that safe...

Thanks for any input,
Cosmin Prund


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RE: [Asterisk-Users] Coice recognition IVR?

[Cosmin Prund]

Thanks!
Can't believe it actually exists...

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Cristian Draghici
 Sent: Monday, April 03, 2006 5:12 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Coice recognition IVR?
 
 Hi there
 
 Sphinx does speech recognition:
 http://www.voip-info.org/wiki-Sphinx
 
 HTH,
 Cristi
 
 On 4/3/06, Cosmin Prund [EMAIL PROTECTED] wrote:
  Hello everyone.
 
  Is it possible to do some very basic voice recognition from within
  Asterisk's dialplan? What I'm aiming at is the ability to speak the
 digits I
  want to dial from my mobile phone. Dialing digits on my mobile phone
 while
  driving is not all that safe...
 
  Thanks for any input,
  Cosmin Prund
 
 
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 --
 Cristian Draghici
 http://www.loudhush.ro
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RE: [Asterisk-Users] Coice recognition IVR?

[Cosmin Prund]

Unfortunately I already gave up myself!

At first glance setting up Sphinx looks like a real pain and, while my
threshold for such pain would definitively allow me to work with it, my
available time can't support this. And I am sorry, because it would look
really nice talking to your box, asking it to reboot or something. Very
star-trek

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Joshua Colp
 Sent: Monday, April 03, 2006 7:06 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Coice recognition IVR?
 
 Cosmin Prund wrote:
  Hello everyone.
 
  Is it possible to do some very basic voice recognition from within
  Asterisk's dialplan? What I'm aiming at is the ability to speak the
 digits I
  want to dial from my mobile phone. Dialing digits on my mobile phone
 while
  driving is not all that safe...
 
  Thanks for any input,
  Cosmin Prund
 
 
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 This has been discussed a lot before, and people usually end up giving
 Sphinx a go and seeing how it is. If you search the mailing list
 archives you might find something useful.
 
 Joshua Colp
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RE: [Asterisk-Users] Problem with System() command.

[Cosmin Prund]

Also be aware Asterisk is probably runing
in its own, non-root account. It needs execute access to the program,
and you need to specify full path. At least thats what worked for me J - dialing 500 on my box
does System(/sbin/reboot) !













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nello Gaudino
Sent: Thursday, March 16, 2006
9:08 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problem
with System() command.







Hi, 
I have an application, script.exe, written under mono
framework and for execute them in my linux box I must write in console: 
mono script.exe 
The problem is that when I call this application in
dialplan with command: 
exten = 500,1,System(mono script.exe) 
the application not run! 
Somebody can help me to find the problem? 
Thanks!










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[Asterisk-Users] What hardware to use for ISDN in Romania

[Cosmin Prund]

Hello everyone.

My land-line provider (Romtelecom) has a very nice offer for ISDN. All in
all they offer me a digital land-line with 1 base number + 2 MSN's and that
would make a grate addition to my full-time home office.

Romtelecom say they're providing EURO-ISDN and the line is compatible with
any euro-isdn compliant equipment. They say they'll install a NT at my
office and this NT will provide me with 1 (one) SO (or was that S0 - zero
opposed to the letter O?) port to connect to my PBX.

My questions:
What hardware do I use to connect the line to my Asterisk?
What are the risks involved (bad drivers etc)?
Has any one used this in Romania?

Thanks!

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RE: [Asterisk-Users] What hardware to use for ISDN in Romania

[Cosmin Prund]

The line is supposed to be standard EUROISDN; I found no mention of DSS1 on
the technical specs page, I only found EUROISDN.

The TELCO is going to provide me with a NT equipment that has two analog
ports and two S0 ports; Of the two S0 ports one is supposed to be used to
connect the PBX to the NT; I've got no idea what the other ports are, I can
only guess the two analog ports will give me access to the two voice
channels using plain-old analog phones.

I'm asking about risks because I ran through the wiki's and ended up very
confused because it seems Asterisk's support for ISDN is driver-dependent
and drivers are obviously kind of hardware dependent.

The risk I'm talking about is signing up for a ISDN contract only to find
I can't get the drivers going, or I can't fully use the service. Since I
don't have access to any other ISDN installation OR ISDN hardware, all I've
got to go on is email, google and the wiki!

As a matter of fact I don't know what hardware to look for! Do I buy this
from a telco provider or from a computer hardware shop? Am I looking for
something listed in the modem category or for some other hardware? Since
there aren't that many Asterisk consultants in Romania I don't really know
where to ask. And yes, I did find MODULO in Bucharest (listed on the wiki as
consultants) but they did not return my last two emails so I'm on my own :-)

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Juergen K. Zick
 Sent: Saturday, March 04, 2006 11:46 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] What hardware to use for ISDN in Romania
 
 HI There
 
 if the line is a standard EUROISDN with DSS1 protocol then it's not a risk
 at all to take and to connect it. You get an S0 interface from your TELCO
 and there cou can plug in any EUROISDN compliant equipment e.g. TAs,
 phones
 and, of course, ASTERISK ...
 
 I would suggest that you AT LEAST try to get an ISDN-TA  (ISDN --PSTN
 converter for old analogue phones and fax machines) as well, for testing
 and backup purposes. They are cheap to get e.g. for abt 2-5 EUR e.g. on
 EBAY.
 
 Depending on your budgets (time and money), experiences and skills you can
 equip your ASTERISK box with incoming and outgoing ISDN channels. You will
 find quite a lot config examples for that, supposingly ISDN-cards with
 HFC-S chipsets are the most versatile ... However, ISDN drivers are still
 a
 bit tricky, but youo have depending on your kernel version at least
 ISDN4LINUX, vISDN, mISDN and chan_modem, chan_capi, chan_capi-cm,
 chan_misdn as config options ...
 
 Anything else you shoul dbe able to find in the WiKis ...
 
 Regards,
 
 Jürgen
 
 
 
 My land-line provider (Romtelecom) has a very nice offer for ISDN. All in
 all they offer me a digital land-line with 1 base number + 2 MSN's and
 that
 would make a grate addition to my full-time home office.
 
 Romtelecom say they're providing EURO-ISDN and the line is compatible
 with
 any euro-isdn compliant equipment. They say they'll install a NT at my
 office and this NT will provide me with 1 (one) SO (or was that S0 - zero
 opposed to the letter O?) port to connect to my PBX.
 
 My questions:
 What hardware do I use to connect the line to my Asterisk?
 What are the risks involved (bad drivers etc)?
 Has any one used this in Romania?
 
 Thanks!
 
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RE: [Asterisk-Users] Changing caller id on transfer

[Cosmin Prund]

Thanks for the tip!
I shoud have found this on my own...

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of C F
 Sent: Friday, March 03, 2006 5:02 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Changing caller id on transfer
 
 Use the following variable in the dialplan to figure out that it has
 been transfered (this only works on a blind transfer) and change CID
 as you wish:
 # ${BLINDTRANSFER}: The active SIP channel that dialed the number.
 This will return the SIP Channel that dialed the number when doing
 blind transfers - see BLINDTRANSFER
 This is a paste from:
 http://www.voip-info.org/wiki-asterisk+variables
 and is also in:
  asterisksource/doc/README.variables and

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RE: [Asterisk-Users] Changing caller id on transfer

[Cosmin Prund]

My dial plan is as simple as it gets:

exten = 101,1,Dial(sip/sip101,180,Ttr)

But I'm doing blind transfers and you're doing attended transfers.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Dinesh Nair
 Sent: Saturday, March 04, 2006 7:05 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Changing caller id on transfer
 
 
 
 On 03/03/06 04:17 Cosmin Prund said the following:
  How can I change the caller id on a transferred call so the called party
  knows the call has been transferred from a colleague and it's not coming
  directly from our outside lines?
 
 ironic ! we're trying to do the reverse:
 
 1. call comes in via our digium zap lines
 2. receptionist answers
 3. receptionist uses atxfer (*1 in features.conf) to transfer to extension
 4. called extension sees callerid of receptionist's extension
 
 we'd like #4 to read, extension called extension sees callerid of
 original
 caller !
 
 could you post your dialplan ?
 
 --
 Regards,   /\_/\   All dogs go to heaven.
 [EMAIL PROTECTED](0 0)http://www.alphaque.com/
 +==oOO--(_)--OOo
 ==+
 | for a in past present future; do
 |
 |   for b in clients employers associates relatives neighbours pets; do
 |
 |   echo The opinions here in no way reflect the opinions of my $a $b.
 |
 | done; done
 |
 +=
 +
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RE: [Asterisk-Users] Polling Asterisk for Life

[Cosmin Prund]

AFAIK there are problems with repeatedly connecting and disconnecting the
manager interface. Also you're probably using a proxy (all manager
interfaces I've seen are using proxies), it might not be a good idea to pool
something out of the manager that often.

Did you consider running a cron job on the server, using asterisk -rx to
run a command and then decide rather asterisk is down or not based on the
result? This way you'd be doing on the server, working around the problems
with the manager interface and saving some bandwidth :). You might also be
able to call /sbin/reboot directly from the cron script!

If on the other hand the whole server is going down you may simply use ping!

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matt
 Sent: Thursday, March 02, 2006 7:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Polling Asterisk for Life
 
 Hi,
 Occassionally Asterisk will go down and I have to restart it.. not
 often.. but sometimes.  When it does the manager interface stops
 working, as does the CLI.
 
 My thoughts was to poll the manager interface once every 5 minutes for
 a value.  If I don't get the value back then alert me that the server
 is possibly down.
 
 Does anyone know what a good value to poll for might be?   I was
 thinking I could poll my SIP account for the CallWaiting value, but
 would like something that was not linked to my account.
 
 Just something that returns a single line is fine.
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[Asterisk-Users] Changing caller id on transfer

[Cosmin Prund]

As usual, this is most likely a easy question, but here it goes any way:

How can I change the caller id on a transferred call so the called party
knows the call has been transferred from a colleague and it's not coming
directly from our outside lines?

The story goes like this:
1) Client calls. All phones ring.
2) Someone picks up the phone.
3) The phone gets transferred to someone.
4) The person that gets the transferred call sees the original caller id and
doesn't know the call has been transferred. I'd like the person that gets
the transfer to see the caller id with a digit prefix. Ex: Original
caller-id: 0269123456; Caller id if the call has been transferred:
1*0269123456

I know I can use SetCallerId(1*${CALLERIDNUM}) but how do I know I'm doing a
transfer and not calling someone?

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RE: [Asterisk-Users] Re: Asterisk at large

[Cosmin Prund]

I'm no Asterisk genie but if you're running on one server you're probably
not dealing with lots of users (for what I'm trying to say 100 users are not
a lot of users). Factoring in the VERY simple format of both sip.conf and
extensions.conf, isn't it possible to create an php page that would
generate those two files from the database? You'll next need to run a basic
script that would call the php's + asterisk -rx reload and you'd be done!

If you're trying to skip the reload step (ie: make the changes available
immediately / transparently) I don't think it can be done, and this has
nothing to do with Asterisk and a lot more to do with databases. Asterisk is
something outside the database, using the database as nothing more but a
source for data. Asterisk will not know the data in the database has
changed, it needs to be told!

On the other hand I am a newbie to Asterisk and I don't really like/know
mySql so I might be very wrong and far from the truth. 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of C F
 Sent: Thursday, March 02, 2006 9:19 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Re: Asterisk at large
 
 Douglas, a lot easier? If it's like you say with multiple servers. But
 the OP did not indicate this in his/her question, in fact s/he sounded
 clueless.
 Also, what is the purpose of NOT having *any* configs from
 /etc/asterisk/
 
 On 3/2/06, Douglas Garstang [EMAIL PROTECTED] wrote:
  Yikes. Managability! It's a lot easier to manage multiple Asterisk
 systems configuration from a single MySQL database then it is to manage
 .conf files on several redundant Asterisk boxes. I can't believe you asked
 that question. I'll apologise in advance because I must be missing part of
 this thread.
 
  -Original Message-
  From: C F [mailto:[EMAIL PROTECTED]
  Sent: Thursday, March 02, 2006 10:16 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Re: Asterisk at large
 
 
  Can you explain why you would want asterisk only thru realtime? and
  not thru the /etc/asterisk/ ?
 
  The wiki is located at:
  http://www.voip-info.org/
  the archives for this list is located at:
  http://lists.digium.com/
  The asterisk irc channel is at:
  irc://irc.freenode.net/#asterisk
  Google is located at:
  http://www.google.com/
  The asterisk docs project is located at:
  http://www.asteriskdocs.org/
 
 
 
 
  On 3/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
   Hi Group,
  
   Please read my previous message below, I want to configure Asterisk
 with Mysql
   and make Asterisk dynamic so that Asterisk will read everything from
 Mysql and
   we can make changes to mysql data directly. Please tell how can we do
 this and
   point me to related documentation.
  
   Thanks for your help and time,
   Manoj.
  
   Quoting [EMAIL PROTECTED]:
  
Hi Group,
   
I was able to install Asterisk and its addons successfully. Now I
 want to
eliminate sip.conf and extensions.conf and use everything from Mysql
 DB, Is
this possible? I have seen this page
   
http://www.voip-
 info.org/wiki/index.php?page=Asterisk%20extensions%20from%20mysql
   
and learnt that we still get the data from Mysql DB and write it as
sub file to
actual sip or extensions.conf before starting Asterisk. Can we
eliminate config
files completely? If it is possible then please point me to the
 links
explaing
how can we do this? I also found very less information on using
 Asterisk with
Mysql, if there are any articles discussing this please send me
 those links.
   
Thanks for your help all the time,
Manoj.
   
  
  
  
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RE: [Asterisk-Users] Changing caller id on transfer

[Cosmin Prund]

I'm doing unattended transfers (ie: I dial #123 to transfer).

I thought there is an easy way to know you're dialing out OR talking to
someone and doing an unattended transfer. If there's no such thing, I'll
just go with the suggestion of prefixing the caller id with something all
phones can understand, and doing this for all calls.

I do not need to care about outbound transfers since all my outbound lines
are FXO and I can't spoof the caller id anyway!

Are there any codes caller-id aware analog phones understand and I can
place in the caller id to be easily identified? 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Joseph Tanner
 Sent: Thursday, March 02, 2006 10:43 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Changing caller id on transfer
 
 Hrm, well it depends exactly how you're transferring calls as to how
 you'd write it in extensions.conf.  Is it being transferred to an
 internal line or to an external line?  If external, then of course you
 need to be able to set the outgoing callerid (you'd basically be
 spoofing it, but that shouldn't be an issue).
 
 I have done something similar, but not exactly like what you're
 wanting.  I'm not sure what the best way to do it would be.  Perhaps
 you could set the callerid early in asterisk in a variable (name it
 something like, ${OUTGOINGCALLERID}).  Before making an outgoing call,
 check asterisk's built-in callerid variable, if it's empty then set it
 to your special variable.  If it's not empty, then use it (so a normal
 outgoing call wouldn't already have callerid set, and would use your
 value, but if an incoming call came in then the callerid variable
 would be set, and we'd use that instead).
 
 The way I did it would require that a user start off in a different
 context based on whether they're receiving a call, or making an
 outgoing call.  Perhaps you can check for a flash, or make them dial a
 special extension to make an outgoing, transferred call?  I dunno, my
 setup's unique and I'm not sure how you can adapt it to your needs.
 Anyways, if you can get them in a different context, then it's simple.
  In your normal outgoing context, the very first line should be what
 sets the callerid.  In the special incoming then outgoing context, do
 something like this:
 
 exten = _1NXXNXX,8,Goto(cell-out,${EXTEN},2)
 
 In this case, _1NXXNXX is the extension matched when I dial a
 normal long-distance number (such as 1-931-555-1212).  It jumps to the
 [cell-out] context (can name this anything you want, this is just my
 setup with calling out via bluetooth), it keeps the extension the same
 (so in [cell-out] we would need an extension of _1NXXNXX), and
 goes to priority 2.  This bypasses the first priority, which is where
 you set callerid for regular outgoing calls, so now you'll use the
 existing value for the outgoing callerid, instead of changing it.
 
 You could just as easily recreate your dialplan for outgoing calls
 that are transferred, but I prefer to jump to an existing context,
 that way I only have to change one part of extensions.conf.  I know
 that if I can make a long-distance call from a local extension, then
 it'll work when someone calls in and gets bridged, because the code is
 exactly the same except for setting callerid.
 
 Hope that helps more than it confuses.
 
 Joseph Tanner
 
 On 3/2/06, Cosmin Prund [EMAIL PROTECTED] wrote:
  As usual, this is most likely a easy question, but here it goes any way:
 
  How can I change the caller id on a transferred call so the called party
  knows the call has been transferred from a colleague and it's not coming
  directly from our outside lines?
 
  The story goes like this:
  1) Client calls. All phones ring.
  2) Someone picks up the phone.
  3) The phone gets transferred to someone.
  4) The person that gets the transferred call sees the original caller id
 and
  doesn't know the call has been transferred. I'd like the person that
 gets
  the transfer to see the caller id with a digit prefix. Ex: Original
  caller-id: 0269123456; Caller id if the call has been transferred:
  1*0269123456
 
  I know I can use SetCallerId(1*${CALLERIDNUM}) but how do I know I'm
 doing a
  transfer and not calling someone?
 
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RE: [Asterisk-Users] Changing caller id on transfer

[Cosmin Prund]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matt Riddell [NZ]
 Sent: Thursday, March 02, 2006 10:55 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Changing caller id on transfer
 
 You could do transfers for a number starting with 8 or whatever
 
 So instead of transferring to 101 (the user's extension), you could
 transfer to 8101.  Then:
 
 exten = _8XXX,1,SetCallerId(1*${CALLERIDNUM})
 exten = _8XXX,2,Goto(extensions,${EXTEN:1},1)

Neh... too much trouble. I'd rather prefix all calls with the internal
flag. Most people would not remamber the extra codes any way.

 Please not that the SetCallerID has been deprecated and should be
 replaced in versions 1.2 with:
 
 Set(CALLERID(number)=1*${CALLERIDNUM})
 
 --
 Cheers,
 
 Matt Riddell

Thanks for the tip!

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RE: [Asterisk-Users] transferring 3000 SIP calls

[Cosmin Prund]

A thread on running 5000 simultaneous cllas ran on this list recently and it
did generate a lot of heat. You might want to look it up the archives - but
make sure you read as many posts on it as possible because lots of different
opinions formulated over time.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Vic
 Sent: Tuesday, February 28, 2006 10:07 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] transferring 3000 SIP calls
 
 Hi, all,
 
 we are building a forwarding station in Japan where we
 would be receiving and forwarding over 3000 SIP calls at
 the same time.
 
 The calls will be offered to us via a carrier as SIP and
 we will forward the call via the same carrier as SIP.
 
 The callflow would look like this:
 
 1. SIP call come in
 2. System will authenticate the call based on the number
 3. Check the billing information and if it is ok, forward
 the call to another number (as SIP)
 4. If call is not ok, system will connect the call to IVR
 for an announcement and touch-tone input
 
 We are thinking about using Asterisk for this.
 How big of a system should it be?
 
 Can we use one linux box for this (and another for backup)
 or will it be something humangously huge?
 
 Thanks,
 Vic
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RE: [Asterisk-Users] fax receive using TDM400P

[Cosmin Prund]

How about this:

--- Results after 33 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.988163

Faxing is working just fine. Mabe it's mother board related?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Anton Krall
 Sent: Tuesday, February 28, 2006 4:41 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] fax receive using TDM400P
 
 Yep, been there, done that.
 
 How about this results:
 
 [EMAIL PROTECTED] asterisk]# /usr/src/zaptel-1.2.4/zttest  -v
 Opened pseudo zap interface, measuring accuracy...
 
 8192 samples in 8191 sample intervals 99.987793%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8191 sample intervals 99.987793%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8193 sample intervals 99.987793%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 8192 samples in 8192 sample intervals 100.00%
 --- Results after 15 passes ---
 Best: 100.00 -- Worst: 99.987793 -- Average: 99.997559
 
 Anything above 99.98 is good so.. Why isnt faxing working :(
 
 |-Original Message-
 |From: [EMAIL PROTECTED]
 |[mailto:[EMAIL PROTECTED] On Behalf Of
 |Rich Adamson
 |Sent: Tuesday, February 28, 2006 2:18 AM
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: RE: [Asterisk-Users] fax receive using TDM400P
 |
 |
 | Ok 1 for Debian, any Fedoras Core 3 out there?
 |
 |fc3, and it doesn't work.
 |
 |If you check the archives, this has all been discussed before.
 |The issue seems to be more oriented to the specific pci bus
 |implementation on the motherboard. You might also want to run
 |/usr/src/zaptel/zttest and read the archives on that as well.
 |
 |
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RE: [Asterisk-Users] fax receive using TDM400P

[Cosmin Prund]

I've noticed some other odd thing with rxfax. In my case I can receive faxes
(using TDM400P) just fine. I can only see those faxes using Windows XP's
Fax and Picture thingy, other applications are having trouble. Also
printing those faxes is a bit odd: the preview is just fine but I always
need to specify landscape printing for portrait faxes. If I print an
portrait fax using potrait setting the fax is actually printed
landscape, shrinked on it's vertical dimension and widend on it's horizantal
dimension. Really funny! I don't know if this is a problem with the viewer
application or with the tiff file itself... 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Anton Krall
 Sent: Saturday, February 25, 2006 3:56 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] fax receive using TDM400P
 
 So, why/how is iaxmodem/hylafax more sucessful in receiveing faxes thru
 tdm
 than rxfax?
 
 I havent been able to get faxes with rxfax, all faxes come in as garbage
 or
 broken or just the first page.
 
 Im hoping and placing my bet on iaxmodem.
 
 |-Original Message-
 |From: [EMAIL PROTECTED]
 |[mailto:[EMAIL PROTECTED] On Behalf Of
 |Administrator TOOTAI
 |Sent: Saturday, February 25, 2006 7:07 AM
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: Re: [Asterisk-Users] fax receive using TDM400P
 |
 |Anton Krall a écrit :
 |
 |Why is iaxmodem with hylafax more stable than spandsp?
 |
 |Can you run iaxmodem and hylafax together with spandsp (for
 |running E1
 |r2mfc)?
 |
 |
 |You're mixing thinks: iaxmodem+hylafax is equivalent to
 |rx_fax/tx_fax, both are based on spandsp which is the library.
 |
 ||-Original Message-
 ||From: [EMAIL PROTECTED]
 ||[mailto:[EMAIL PROTECTED] On Behalf
 |Of Rob Danz
 ||Sent: Friday, February 24, 2006 9:34 AM
 ||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 ||Subject: RE: [Asterisk-Users] fax receive using TDM400P
 ||
 ||I wrestled with this for a long time, as have many others
 |and it just
 ||doesn't work with spandsp and asterisk alone.
 ||
 ||Use iaxmodem and hylafax in conjunction with asterisk... it
 |works like
 ||a champ.  I have a single POTS line coming in so I get voice  fax
 ||with a single number using fax detect.
 ||
 ||http://iaxmodem.sourceforge.net/
 ||
 ||
 ||
 ||-Original Message-
 ||From: Rich Adamson [mailto:[EMAIL PROTECTED]
 ||Sent: Friday, February 24, 2006 7:28 AM
 ||To: Asterisk Users Mailing List - Non-Commercial Discussion
 ||Subject: Re: [Asterisk-Users] fax receive using TDM400P
 ||
 || Ive been testing how to receive faxes using TDM400P cards and so
 || far,
 ||after
 || playing with gains, echocancell and echotraining on
 ||zapata.conf.. Ive
 || ha
 ||dno
 || luck, faxes come in as garbage or broken or with blank lines.
 ||
 || Anybody has successfully done this? Any tips.. Also I have
 ||some ideas:
 ||
 || 1. Is it really possible to get fxes on a fax machine using
 ||ATAs like
 || the sipura 2002? Even using ulaw and pass-thru, is it possible?
 ||
 || 2. Since the faxes is coming from PSTN thru the card, I
 ||guess asterisk
 ||will
 || always stay in the middle right? No way around this.
 ||
 || 3. Im also trying to receive faxes usign a TE110P card
 |with spandsp,
 ||unicall
 || and E1 R2MFC, no luck also, some stuff, garbage and broken faxes.
 || Anybody done this sucessfuly?
 ||
 || Hope anybody can share their thoughts and insight on this.
 ||
 ||Using the TDM400 card for any form of fax'ing (or modem use) is well
 ||known to be unreliable and, in most cases, totally unusable.
 |The issue
 ||has been discussed many times over the last two years or so.
 |There are
 ||no known workarounds.
 ||
 ||Its my understanding that lots of folks have spandsp working via T1
 ||and/or PRI interfaces. The issues associated with the TDM400 card do
 ||not apply to the T1 cards.
 ||
 ||
 ||
 ||
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RE: [Asterisk-Users] fax receive using TDM400P

[Cosmin Prund]

I know some (most?) of you will say this is wrong but...
When using rxfax my faxes get generated in a /fax folder and that folder is
shared using samba :-) It works sooo nice! If I could only get myself to
trust rxfax so I can free the FXS port for some other duty!

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Thomas Artner
 Sent: Saturday, February 25, 2006 8:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] fax receive using TDM400P
 
 Am Saturday 25 February 2006 19:38 schrieb Steve Underwood:
  Cosmin Prund wrote:
  I've noticed some other odd thing with rxfax. In my case I can receive
   faxes (using TDM400P) just fine. I can only see those faxes using
 Windows
   XP's Fax and Picture thingy, other applications are having trouble.
   Also printing those faxes is a bit odd: the preview is just fine but I
   always need to specify landscape printing for portrait faxes. If I
   print an portrait fax using potrait setting the fax is actually
   printed landscape, shrinked on it's vertical dimension and widend on
 it's
   horizantal dimension. Really funny! I don't know if this is a problem
   with the viewer application or with the tiff file itself...
 
  Its the viewers. A large number of TIFF viewers are badly broken. Some
  only show the first page. Some do not obey the standard/fine resolution
  things properly, and get things very squashed.
 
 i think the better way is to convert the tiff to pdf before sending the
 file
 to the enduser!
 
 
 
 tom
 
 
  Steve
 
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RE: [Asterisk-Users] fax receive using TDM400P

[Cosmin Prund]

Windoes'es DEFAULT image viewer also has problems showing rxfax-generated
TIFF's. They do show up properly on screen but when printed the orientation
needs to be changed. I don't know if MS's viewer is somehow broken OR the
tiff is somehow broken (also I don't care) but viewing rxfax-ed tiffs is not
what you see is what you get.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Steve Underwood
 Sent: Saturday, February 25, 2006 9:11 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] fax receive using TDM400P
 
 Thomas Artner wrote:

 The funny thing most people are using Windows, and only have trouble
 with this because they replace the default Windows viewer with something
 broken. The default viewer may look dull and boring, but its one MS
 component that actually does a decent job.
 
 Using PDFs is one solution. The number of readers is very small, and in
 the Windows world there is pretty much only one. You know a PDF will be
 seen with one of a small number if reader, all of which do a fair job.
 Every fool seems to think they can cook up a uniquely wonderful image
 viewer, and other fools keep installing them.
 
 Regards,
 Steve
 
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[Asterisk-Users] Detecting disconnect on TDM400P with 3 FXO ports and 1 FXS port

[Cosmin Prund]

Hellow everyone, here's an other newby question.

I've got a * configured with the card in the subject line. At times Asterisk
fails to notice a disconet from the incoming line going into one of the FXO
ports. Consequently it just keeps the line off-hook for ever and that causes
my provider to mark the line aut of order.

Is there any way to help Asterisk notice the disconect?

This are the relevant parts of my zapata.conf:

Callwaiting=no
Usecallingpres=yes
Callwaitingcallerid=yes
Threewaycalling=no
Transfer=yes
Cancallforward=yes
Callreturn=yes
Echocancel=yes
Echocancewhenbridged=no
Echotraining=800
Rxgain=0.0
Txgain=0.0
Group=0
Callgroup=1
Pickupgroup=1
Faxdetect=incoming
Immediate=yes
Signaling=fxs_ks
Context=from_rtc
Busydetect=yes

Channel = 4

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RE: [Asterisk-Users] Intro and first questions

[Cosmin Prund]

I'm a newbye myself so beware!

(1) http://www.voip-info.org is your friend. I've got most of my info off
that site and it's a good place to start.

(2) Download a Softphone like XLite (you'll also find info on softphones on
voip-info) and start experimenting on site. When you'll be able to
configure your softphone to call an other softphone on a different machine
you'll be on your way to setting up the link with your daughter.

(3) If you've got a bit of experience with Linux and it's style of
configuration files stay away from automated GUI's like AMP and stuff as
they add an other level of abstraction on top of an already complex thing.
Resolving the issues that you'll probably run into will be a lot easier if
you typed the whole configuration files your self (as opposed to having them
generated by things like AMP). Out of my experience, after staring with a
fresh install of [EMAIL PROTECTED] I had to basically DELETE everything in my
extensions.conf (the dial plan) as I was unable to make any sense of it. It
was a complex thing generated by AMP. I'm sure it was much better then my
own but I was plain simply unable to understand it!

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Tom Poe
 Sent: Saturday, February 18, 2006 6:08 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Intro and first questions
 
 This is my first venture into VoIP from my Fedora Core 4 system.  I came
 across a posting:
 
 http://atrpms.net/name/asterisk/
  http://atrpms.net/name/asterisk-addons/
  http://atrpms.net/name/asterisk-sounds/
  http://atrpms.net/name/spandsp/
  http://atrpms.net/name/libpri/
  http://atrpms.net/name/zaptel/
 on the Fedora list, added the repository to yum, and downloaded,
 installed, then typed:
 # asterisk -c , hit the return,
 and a bunch of stuff happened, before returning to the root prompt.
 
 My first goal(s) is to be able to configure the machine to make a PC to PC
 call to my daughter, who lives in Minnesota.  If all goes well, I can set
 up her computer to receive the call, using Asterisk.  Is this a realistic
 first experience project?  If so, is there a tutorial out there that
 describes the steps I need to take?  Any advice, suggestions, greatly
 appreciated.
 Tom
 
 
 --
 94% of returning troops suffer from trauma
 Open Studios
 http://www.ibiblio.org/studioforrecording/
 
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