RE: [asterisk-users] Rxfax and Txfax on Asterisk 1.4

[Ardjan Zwartjes]

>It's not a timing problem, it's a "that just happens when you do faxing

>over typical VoIP channels" problem.  The reason it happens is because 
>of jitter... and more specifically, packet loss.  On UDP communications

>packets are not retransmitted if they get dropped by a router or switch

>or hub.  You can't just take a data stream and chop out small chunks of

>it here and there and expect the receiving end to be okay with it.

I think you missed my point here, I understand the risks of faxing over
VOIP but our problem ONLY occurs when then entire path of the fax is in
voip. If we use a 10 times longer VOIP path combined with a small path
over a regular telephony channel everything works fine, even though this
would still suffer from packetloss. You're problably right about the
jitter part (which is a timing problem as far as I can see) and I expect
that the regular telephony channel fixes the problem since it has strict
timing characteristics.

Kind regards,
Ardjan Zwartjes.
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Re: [asterisk-users] Problems with GXP2000 and Asterisk => Call pickup and Voicemail

[Gordon Henderson]

On Fri, 9 Feb 2007, Noc Phibee wrote:


Hi

i have two problems with my Grandstream GXP2000 :

  1- When i wan pickup a call, that's don't work's (*8EXTEN)
   and when i test whit Softphone, i have a error too, he say me
  [EMAIL PROTECTED] not found ..
  in features.conf, i have:

[general]
  parkext => 700parkpos => 701-720
  context => parkedcalls
  pickupexten = *8


I'm under the impression that *8 picks up any ringing phone in the same 
group... Not sure why youre dialling an extension number after it... I may 
be wrong though - I've never used it!




  2- When i want access to the voice server, he never understand my
  password ... but with a softphone that's work's


Anyone have this problems too ?


I'd guess that asterisk isn't hearing the tones of the password?

Start with putting

   dtmfmode=rfc2833

in your sip.conf file, and making that setting on the GPX2000 phone 
itself (on the account page)


Gordon
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RE: [asterisk-users] TDM400 with 1 FXO

[[EMAIL PROTECTED]]

Try to do "lspci" to see if your card has been found by linux os.
   
  And then,  try the command "lsmod" to see if the zaptel module and wctdm 
module have been loaded.
  

"Klaverstyn, David C" <[EMAIL PROTECTED]> wrote:
  When I made a change I was always doing a ztcfg -vv and then an asterisk
reload.

I have rebooted the server now but still have the same problem.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, 9 February 2007 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO

Klaverstyn, David C wrote:
> My original post does have the contents of the file exactly.
>
> In my /etc/asterisk/zapata.conf file I have 
> [trunkgroups]
>
> [channels]
> context=from-pstn
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
>
> 
You need to insert signalling before the channel statement:
signalling=fxs_ks
> channel => 1
>
> 
>> Yes, I have also since put that in and I get the error:
>> Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
>> signalling
>> 
This warning happens when you change the signalling in zapata.conf 
without restarting asterisk. I suspect you did a zap reload on the
console.

I'd suggest you restart both zaptel and asterisk to keep everything in
sync.

Leo
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=

Provide IAX2 termination for VOIP beginners and individual end-users.  
No minimum usage. 
Register now and get free $0.2 credit.

http://www.unifycall.com

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[asterisk-users] call park and call transfer example

[Rilawich Ango]

Hi all,

 Does any can give me some example to setup call parking and call
transfer of a call?
In my understanding, call parking and call transfer should be like
something below.  Am I right?

ango


Call parking:
caller A -> callee B
callee B park her call
callee B get back her call in another phone

Call transfer:
caller A -> callee B
callee B transfer to C
finally: A talks to C
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RE: [asterisk-users] Any Way to Get # Functionality in DISA

[Yuan LIU]

From: "Yuan LIU" <[EMAIL PROTECTED]>
Date: Thu, 08 Feb 2007 21:28:03 -0800
Not necessarily.  You only have to program your existing context to handle 
trailing # when it comes along.  For example, this simplistic example 
ignores trailing #'s:


exten => _Z., 1, GotoIf($[${EXTEN:-1} = #]?${EXTEN:1},1:2)
exten => _Z., 2, whatever...


Or simply add

exten => _Z.[#*].,1, Goto($[${EXTEN} : \([0-9]*\)],1)

to an existing context to ignore anything after first # or *.


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Re: [asterisk-users] does any one knows of a Softphone that worksunder terminal

[Yuan LIU]

From: Manrique Feoli <[EMAIL PROTECTED]>
Date: Thu, 08 Feb 2007 15:46:11 -0600

I wonder if I setup a softphone on each terminal if they will actually work 
as independent phones well enough, but haven´t tested it.


You mean simultaneously?  Port conflict might be one reason why it won't, 
unless you are willing to configure each login's softphone manually.  I'd 
rather go with VMWare or VirtualPC if I have to. Even there, transmitting 
audio via RDP could add another layer of unreliability.


What's so wrong about embedding a Java soft phone applet, or even 
distributing a Java Web application?


Talking off my hat.  No real world experience.

Yuan Liu


MF escribió:

Hi all

I'm looking for a softphone that works well under terminal services 
environment,


we need to set up  24 to 32 phones for a call center,

also, does any one knows if it  will actually work fine under load?



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RE: [asterisk-users] Any Way to Get # Functionality in DISA

[Yuan LIU]

From: "Robert DeVries" <[EMAIL PROTECTED]>
Date: Thu, 8 Feb 2007 13:09:06 -0800

When using a SIP phone with Asterisk, hitting the # key (pound or hash
depending on where in the world you happen to be) tells Asterisk that there
are no more digits coming, and to put the call through immediately based on
the digits already entered.  This is the same functionality as the PSTN (at
least in North America).


Not quite.  This behaviour is more than likely a configurable feature in a 
hard SIP phone. (At least in my GrandStream it is.)  When you dial into 
PSTN, it doesn't matter what else you punch after matching the dial plans 
your telco configured at their end (NANP in North America, usually 7- and 
10-digit; cell phones do 9-digit because there is no concept of "dial", only 
"send").  # is used as a customary terminator only in some IVR systems. 
(Banking, calling card, ...)



However, DISA just sees the # as another digit, and therefore pressing #
produces an error.   I suppose it would be possible to write something in
the dial plan that looks for a # then strips the # and puts the call
through, but I'm curious if there is any other way to do this.  For one
thing, this would require having a special context for DISA, rather than
just having DISA place calls through an existing context.


Not necessarily.  You only have to program your existing context to handle 
trailing # when it comes along.  For example, this simplistic example 
ignores trailing #'s:


exten => _Z., 1, GotoIf($[${EXTEN:-1} = #]?${EXTEN:1},1:2)
exten => _Z., 2, whatever...

You can use regular expressions to do more sophisticated handling.  Note 
this example will not emulate a common IVR convention of stop reading when # 
is pressed; that one is implemented in Read().  You'll have to decide 
whether Read() is more appropriate for your application. (Read() will force 
users to press #, I think.)


Hope this helps.

Yuan Liu


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Re: [asterisk-users] requesting real world meetme capacity numbers

[Paul Hales]

Above a certain number, background noise destroys the conference.

Anyone remember what the number is?

PaulH

On Thu, 2007-02-08 at 22:31 -0600, JR Richardson wrote:
> Hi All,
> 
> I'm very interested in real world experience of double digit number of
> users sustaining good quality audio in a single meetme conference.
> 
> Personally, I have seen 23 users in one conf room, all coming in SIP,
> ULAW.  Server is 3.2GHz proc, 1Gig RAM, 1-2 % proc utilization under
> 23 user load, perfect audio.
> 
> I'm working on a conf bridge for 150+ users, could use some advice, if
> anyone has accomplished such a feat or has any ideas on how.
> 
> Thanks.
> 
> JR
> 

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RE: [asterisk-users] TDM400 with 1 FXO

[Yuan LIU]

From: "Klaverstyn, David C" <[EMAIL PROTECTED]>
Date: Fri, 9 Feb 2007 13:30:02 +1100

Yes, I have also since put that in and I get the error:
Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
signalling


This is not quite an error.  Did you do a reload or a fresh start?  I get 
this warning every time I reload.  Have you tried 'show modules like chan_' 
or 'show channeltypes' and confirm that chan_zap is not loaded after putting 
in signalling=fxs_ks (not fxsks like in zaptel.conf)?


If anything, test if wctdm is indeed configured correctly using zttool 
(under your compiled source tree).


Yuan Liu


And if I put in rxwink I get this error:
Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
rxwink

It's all very strange.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, 9 February 2007 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO

Klaverstyn, David C wrote:
> Hi,
>
> Yes it should, I have changed it back and is still causing the same
> problems.
>
Did you also missed out the following line in zapata.conf?
signalling=fxs_ks

Leo



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[asterisk-users] Problems with GXP2000 and Asterisk => Call pickup and Voicemail

[Noc Phibee]

Hi

i have two problems with my Grandstream GXP2000 :

   1- When i wan pickup a call, that's don't work's (*8EXTEN)
and when i test whit Softphone, i have a error too, he say me
   [EMAIL PROTECTED] not found ..
   in features.conf, i have:

 [general]
   parkext => 700 
   parkpos => 701-720

   context => parkedcalls
   pickupexten = *8


   2- When i want access to the voice server, he never understand my
   password ... but with a softphone that's work's


Anyone have this problems too ?

Thanks bye
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RE: [asterisk-users] TDM400 with 1 FXO

[Klaverstyn, David C]

When I made a change I was always doing a ztcfg -vv and then an asterisk
reload.

I have rebooted the server now but still have the same problem.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, 9 February 2007 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO

Klaverstyn, David C wrote:
> My original post does have the contents of the file exactly.
>
> In my /etc/asterisk/zapata.conf file I have 
> [trunkgroups]
>
> [channels]
> context=from-pstn
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
>
>   
You need to insert signalling before the channel statement:
signalling=fxs_ks
> channel => 1
>
>   
>> Yes, I have also since put that in and I get the error:
>> Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
>> signalling
>> 
This warning happens when you change the signalling in zapata.conf 
without restarting asterisk. I suspect you did a zap reload on the
console.

I'd suggest you restart both zaptel and asterisk to keep everything in
sync.

Leo
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Re: [asterisk-users] Skutch AS-66 and an X100P

[Leo Ann Boon]

Yuan LIU wrote:


Kind of do.  There are times when it feels like trying to fit two 
spinning wheels, though:-)
'Zee trick to fit two spinning wheels is to stop the wheels :)'. That 
why, your first working system is the most important. It's easier to 
built on once you have a solid foundation. Everyone has to go through 
this rite of passage.


Leo

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Re: [asterisk-users] TDM400 with 1 FXO

[Leo Ann Boon]

Klaverstyn, David C wrote:

My original post does have the contents of the file exactly.

In my /etc/asterisk/zapata.conf file I have 
[trunkgroups]


[channels]
context=from-pstn
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

  

You need to insert signalling before the channel statement:
signalling=fxs_ks

channel => 1

  

Yes, I have also since put that in and I get the error:
Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
signalling

This warning happens when you change the signalling in zapata.conf 
without restarting asterisk. I suspect you did a zap reload on the console.


I'd suggest you restart both zaptel and asterisk to keep everything in sync.

Leo
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RE: [asterisk-users] TDM400 with 1 FXO

[Klaverstyn, David C]

Hi,  I stuffed Up,

Here is my correct contents


; Configuration file
 
[trunkgroups]
 
[channels]
 
language=en
context=from-pstn
signalling=fxs_ks
rxwink=300   ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
 
busydetect=yes
busycount=6
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
 
channel => 1


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Friday, 9 February 2007 2:29 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] TDM400 with 1 FXO

>From: "Klaverstyn, David C" <[EMAIL PROTECTED]>
>Date: Fri, 9 Feb 2007 15:12:49 +1100
>
>My original post does have the contents of the file exactly.

You haven't defined channel signaling in zapata.conf.  Need something
like
signalling = fxs_ks
according to your zaptel.conf.

Yuan Liu

>In my /etc/asterisk/zapata.conf file I have
>[trunkgroups]
>
>[channels]
>context=from-pstn
>usecallerid=yes
>hidecallerid=no
>callwaiting=yes
>usecallingpres=yes
>callwaitingcallerid=yes
>threewaycalling=yes
>transfer=yes
>canpark=yes
>cancallforward=yes
>callreturn=yes
>echocancel=yes
>echocancelwhenbridged=yes
>rxgain=0.0
>txgain=0.0
>group=1
>callgroup=1
>pickupgroup=1
>immediate=no
>
>channel => 1
>
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
>Boon
>Sent: Friday, 9 February 2007 12:44 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] TDM400 with 1 FXO
>
>Klaverstyn, David C wrote:
> > Yes, I have also since put that in and I get the error:
> > Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
> > signalling
> >
> > And if I put in rxwink I get this error:
> > Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
> > rxwink
> >
> > It's all very strange.
> >
>please post your complete zapata.conf - I think there's a preceding
line
>
>that's confusing the parser.
>
>Leo
>
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[asterisk-users] requesting real world meetme capacity numbers

[JR Richardson]

Hi All,

I'm very interested in real world experience of double digit number of
users sustaining good quality audio in a single meetme conference.

Personally, I have seen 23 users in one conf room, all coming in SIP,
ULAW.  Server is 3.2GHz proc, 1Gig RAM, 1-2 % proc utilization under
23 user load, perfect audio.

I'm working on a conf bridge for 150+ users, could use some advice, if
anyone has accomplished such a feat or has any ideas on how.

Thanks.

JR

--
JR Richardson
Engineering for the Masses
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RE: [asterisk-users] TDM400 with 1 FXO

[Yuan LIU]

From: "Klaverstyn, David C" <[EMAIL PROTECTED]>
Date: Fri, 9 Feb 2007 15:12:49 +1100

My original post does have the contents of the file exactly.


You haven't defined channel signaling in zapata.conf.  Need something like
signalling = fxs_ks
according to your zaptel.conf.

Yuan Liu


In my /etc/asterisk/zapata.conf file I have
[trunkgroups]

[channels]
context=from-pstn
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

channel => 1


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, 9 February 2007 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO

Klaverstyn, David C wrote:
> Yes, I have also since put that in and I get the error:
> Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
> signalling
>
> And if I put in rxwink I get this error:
> Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
> rxwink
>
> It's all very strange.
>
please post your complete zapata.conf - I think there's a preceding line

that's confusing the parser.

Leo

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Re: [asterisk-users] Skutch AS-66 and an X100P

[Yuan LIU]

From: Leo Ann Boon <[EMAIL PROTECTED]>
Date: Fri, 09 Feb 2007 09:12:38 +0800

I don't know anything about a line simulator but your description 
certainly points to a problem with the simulator.  As I'm also doing tests 
on X100P, I'm interested to know what does a simulator give you that your 
PBX doesn't. (I wish I had a PBX to play with.)
How about just using a working Asterisk PBX :)? Or use a good fxs gateway 
that allows you to configure custom tones.


Leo


Kind of do.  There are times when it feels like trying to fit two spinning 
wheels, though:-)


Yuan Liu


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RE: [asterisk-users] TDM400 with 1 FXO

[Klaverstyn, David C]

My original post does have the contents of the file exactly.

In my /etc/asterisk/zapata.conf file I have 
[trunkgroups]

[channels]
context=from-pstn
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

channel => 1


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, 9 February 2007 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO

Klaverstyn, David C wrote:
> Yes, I have also since put that in and I get the error:
> Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
> signalling
>
> And if I put in rxwink I get this error:
> Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
> rxwink
>
> It's all very strange.
>   
please post your complete zapata.conf - I think there's a preceding line

that's confusing the parser.

Leo

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[asterisk-users] Queue extension issues

[John Breen]

I'm stuck on queues!

The way I read what documentation I have found, if I set up a queue like 
this:


[general]
persistentmembers = yes

[testq]
musiconhold=default
strategy = ringall
timeout = 10
retry = 5
context = testing
member => SIP/100


and then add into extensions something like this:

[incomingiax]
exten => 1234,1,Dial(SIP/100,10)
exten => 1234,2,Queue(testq|tTH|||300)

[testing]
exten => 1,1,Dial(SIP/101)

[testcontext]
exten => 100,1,Dial(SIP/100)
exten => 100,hint,SIP/100

exten => 101,1,Dial(SIP/101)
exten => 101,hint,SIP/101

exten => 102,1,Dial(SIP/102)
exten => 102,hint,SIP/102


Then if a user dials in on extension 1234 (which is what's forwarded 
from the iax peer), Ext.100 should ring for
10 sec, then the call should be placed in queue testq.  While the call 
is in the queue , the caller should be able to press 1, which should 
then send them on to ext. 101.  That's right, isn't it?


Problem is, dialling 1 doesn't go to ext. 101

I can't see anything obvious that I've done wrong - It all looks right 
to me.  But I've obviously missed something.  Can anyone enlighten me as 
to what that something might be?


Regards

John Breen
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Re: [asterisk-users] TDM400 with 1 FXO

[Leo Ann Boon]

Klaverstyn, David C wrote:

Yes, I have also since put that in and I get the error:
Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
signalling

And if I put in rxwink I get this error:
Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
rxwink

It's all very strange.
  
please post your complete zapata.conf - I think there's a preceding line 
that's confusing the parser.


Leo

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Re: [asterisk-users] Red alarms

[Wayne Jensen]

On 2/8/07, Don Pobanz <[EMAIL PROTECTED]> wrote:

> Asterisk is getting red alarms on my T1, sometimes once or twice a
> day, but today it happened 5 times.  Even once is too many.  Every
> call in progress is dropped.

Red alarm means that the hardware is not seeing the T1 signal coming in.
This most likely is a cable or wiring or perhaps a hardware problem.



I know that it's not wiring because it's never happened before in the
year and a half that we've been using it, and if I disconnect the
asterisk box and use it the same way we've been using it, the problem
goes away.  same cabling, just unplugging the cable from the channel
bank and plugging it into the Digium card.

does this mean that the Digium card is bad?  or is there something in
the configuration that could make it not see the T1 signal coming in?
I don't see how it would be, but is it possible that a bad cable
between the Digium card and the channel bank (channels 49-72 on the
card) could cause a red alarm on channels 1-24?

I've used the same Digium card in two different boxes and got the same
red alarms, so I doubt it's a problem with the computer it's in.

The red alarms only happen when there's a high volume of calls going
through, but it doesn't *always* happen when there's a high volume.
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RE: [asterisk-users] TDM400 with 1 FXO

[Klaverstyn, David C]

Yes, I have also since put that in and I get the error:
Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
signalling

And if I put in rxwink I get this error:
Feb  8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
rxwink

It's all very strange.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, 9 February 2007 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO

Klaverstyn, David C wrote:
> Hi,
>
> Yes it should, I have changed it back and is still causing the same
> problems.
>   
Did you also missed out the following line in zapata.conf?
signalling=fxs_ks

Leo

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RE: [asterisk-users] TDM400 with 1 FXO

[MBIT Technologies]

Hi David

Also make sure the power connector is also connected to the board.
 

Regards
 
 
Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 4950 5609
E: [EMAIL PROTECTED]
W: http://www.mbit.com.au
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn,
David C
Sent: Friday, 9 February 2007 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] TDM400 with 1 FXO

Hi,

Yes it should, I have changed it back and is still causing the same
problems.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, 9 February 2007 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO


Klaverstyn, David C wrote:
>
> Hi All,
>
>  
>
> I cannot get my TDM to work correctly.
>
>  
>
> In my /etc/zaptel.conf file I have
>
> loadzone = us
>
> defaultzone=us
>
>  
>
> fxoks=1
>
Shouldn't this be fxsks if you're using an FXO module as analog trunk?

Leo

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Re: [asterisk-users] TDM400 with 1 FXO

[Leo Ann Boon]

Klaverstyn, David C wrote:

Hi,

Yes it should, I have changed it back and is still causing the same
problems.
  

Did you also missed out the following line in zapata.conf?
signalling=fxs_ks

Leo

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Re: [asterisk-users] Asterisk outbound calling does not wait for answer before playback

[Leo Ann Boon]

Alyed Tzompa wrote:
Had the same issue time ago, but Eric shed good light on it, have a 
look at:


http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html

Summary: sorry, no nice work around.
At least, not in the analog TDM world. Personally, I'll advise everyone 
to use ISDN if you need to detect call progress for TDM circuits. Or if 
you're in North America, try the callprogress=yes option in zapata.conf.


Analog lines are perfectly fine is you're happy with call progress 
detection in Wetware(TM) :). For automated call progress detection, you 
would be happier with a digital line or VOIP.


Leo


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[asterisk-users] SIP Re-Invite behind a NAT

[hugolivude]

SetUp:
- Asterisk behind a NAT,
- Red Hat 9.0
- Asterisk 1.2.14

My Asterisk box is behind a NAT and I have a DiD from an ITSP.  I have
my dial plan set up so that when outside callers  dial the DiD, the
call is answered by my auto-attendant.  The caller can then select who
they'd like to speak to and the call is transferred to the  external
line associated with that person (usually a mobile phone) using a
Dial() command.

Because both parties are external, I don't want the media to pass
through my Asterisk box once the two parties connect. This is the
default behaviour for the Dial command if you avoid the tT options and
the codec is supported all the way through - this is true in my case.

I have this working great with IAX - I can even disconnect the
Ethernet cable from my Asterisk box once the call is established and
the call is not affected.  Unfortunately I cannot get it to work with
SIP and my ITSP is dropping support for IAX.

Can you help me?

I've attached the relevant bits from sip.conf.  I have canreinvite=yes
and I've tried with nat=yes and nat=no, but no luck either way - the
call goes through in each case but the media is passing through my
Asterisk box and i'd like to avoid that.

Anxiously awaiting a reply.

Thanks,
H


register => me: [EMAIL PROTECTED]:5060
register => me: [EMAIL PROTECTED]:5060


; This section is because i'm behind nat
;
externip=999.999.999.999 ;Outside address
localnet= 192.168.0.148/255.255.255.0 ;Inside Network
;
;
;
[voip-ITSP1]
context=incoming-sip
type=friend
host= my1.itsp.com
username=me
secret=xxx
nat=no
canreinvite=yes
insecure=port,invite ; do NOT remove this
qualify=yes ; do NOT remove this
dtmfmode=rfc2833 ; should match what is set on your account
disallow=all
allow=ulaw ; set in/out codec here
;
[voip-ITSP2]
context=incoming-sip
type=friend
host= my2.itsp.com
username=me
secret=xxx
nat=no
canreinvite=yes
insecure=port,invite ; do NOT remove this
qualify=yes ; do NOT remove this
dtmfmode=rfc2833 ; should match what is set on your account
disallow=all
allow=ulaw ; set in/out codec here
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RE: [asterisk-users] TDM400 with 1 FXO

[Klaverstyn, David C]

Hi,

Yes it should, I have changed it back and is still causing the same
problems.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, 9 February 2007 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO


Klaverstyn, David C wrote:
>
> Hi All,
>
>  
>
> I cannot get my TDM to work correctly.
>
>  
>
> In my /etc/zaptel.conf file I have
>
> loadzone = us
>
> defaultzone=us
>
>  
>
> fxoks=1
>
Shouldn't this be fxsks if you're using an FXO module as analog trunk?

Leo

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Re: [asterisk-users] Re: Auto Answer (Paging)

[Shane Spencer]

I hate to say this, but voip-info.org has a few different methods of
handling this already defined.

If you are 'intercomming' to several styles of SIP based phones, you
have but to only configure the phone to accept those types of calls
and add a SIP header pre Dial().

Shane
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Re: [asterisk-users] TDM400 with 1 FXO

[Leo Ann Boon]

Klaverstyn, David C wrote:


Hi All,

 


I cannot get my TDM to work correctly.

 


In my /etc/zaptel.conf file I have

loadzone = us

defaultzone=us

 


fxoks=1


Shouldn't this be fxsks if you're using an FXO module as analog trunk?

Leo

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[asterisk-users] TDM400 with 1 FXO

[Klaverstyn, David C]

Hi All,

 

I cannot get my TDM to work correctly.

 

In my /etc/zaptel.conf file I have

loadzone = us

defaultzone=us

 

fxoks=1

 

 

In my /etc/asterisk/zapata.conf file I have 

[trunkgroups]

 

[channels]

context=from-pstn

usecallerid=yes

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

canpark=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=yes

rxgain=0.0

txgain=0.0

group=1

callgroup=1

pickupgroup=1

immediate=no

 

channel => 1

 

 

 

 

running a ztcfg -vv give the following

Zaptel Configuration

==

Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)

1 channels configured.

 

 

 

 

A reload of asterisk give the following error:

Feb  8 18:38:43 ERROR[4558]: chan_zap.c:10305 setup_zap: Unable to
reconfigure channel '1'

Feb  8 18:38:43 WARNING[4558]: chan_zap.c:11067 reload: Reload of
chan_zap.so is unsuccessful!

 

 

Any help would be greatly appreciated.  For what it is worth this system
is located in Canada.

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Re: [asterisk-users] Skutch AS-66 and an X100P

[Leo Ann Boon]

I don't know anything about a line simulator but your description 
certainly points to a problem with the simulator.  As I'm also doing 
tests on X100P, I'm interested to know what does a simulator give you 
that your PBX doesn't. (I wish I had a PBX to play with.)
How about just using a working Asterisk PBX :)? Or use a good fxs 
gateway that allows you to configure custom tones.


Leo

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[asterisk-users] Help - Poor Voice Quality

[Gary G. Hendershot]

I have been with Teliax for some time ... Overall I like the service very
much ... The key things offered that got me to go with them (having been
with VoicePulse and BroadVoice) was live person in tech support and choice
of either SIP or IAX trunking ( I have 4xCalls with the plan I bought ) ...

However, I also had sound quality issues with them early on when I used IAX
... As soon as I switched to SIP, the problem went away and the quality was
outstanding ...

Also, the New York gateway you are using was still in BETA last time I
checked ... You might want to try using the voip-co1.teliax.com gateway with
SIP ... See how that works ... 

If not, give the tech support guys at Teliax a shot ... I have found them to
be quite responsive the few times I have needed them ...

G.Hendershot


-Original Message-
From: Jim Duda [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, February 06, 2007 8:10 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Help - Poor Voice Quality

I'm struggling to get my VOIP installation to be acceptable.  I'm looking
for advice on what else I can look for.

My system:
o Teliax VOIP service, voip-ny1 proxy
o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average
jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server
lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with
QOS configured for VOIP P1 o Asterisk 1.4.0 o Sipura SPA-2000 o Grandstream
GXP-2000 o IAX connection to teliax

Outbound voice quality is many times horrible, to the point where ppl say
they cannot hear me.  The voice often drops out.  Inbound quality seems to
cut in and out too.

I downloaded the myVoipSpeed VOIP analyzer.  It indicates that I have plenty
of download and upload bandwidth.  I also have good jitter.  The tool
doeesn't find any packet loss whatsoever.

My RCN cable company cannot find anything wrong with my cable modem.  No
packet loss.  I'm supposed to be paying for 10M bit downloads, but only
getting 3M bit.

I've been on the shorewall firewall and confirmed that I have the firewall
configured properly for VOIP QOS.

I'm using the basic asterisk iax.conf setup with only those changes required
to interface with the teliax service.

I have the same issues with both the Sipura Adapter and the Grandstream
phones, however, I do believe the Grandstream appears worse at times.

I've attempted to analyze the IAX traffic using the Wireshark ethernet
protocol analyzer.  Everything looks okay best I can tell.

What else can I do to analyze why the voice quality is so bad?
What can I do in Asterisk to help track down where the problem is?

I want to make this VOIP work.

Thanks for any help.

Jim



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[asterisk-users] Re: Auto Answer (Paging)

[Justin Newman]

What kind of phones? Polycom? You could also use one of the sip-based speakers 
or intercom units...

--

From: Rob Schall <[EMAIL PROTECTED]>
Subject: [asterisk-users] Auto Answer (Paging)

I'm trying to duplicate a behavior we had with our old avaya system, and
I've come across Auto Answer (Ring Answer). However, its not quite the
same yet.

Right now, when I dial **5053, it will add the SIP header for Ring
Answer and it will call 5053. The phone auto pickups just fine. However,
we need that call to be muted. If you were to call into a meeting, we
wouldn't want them to hear that meeting, but instead the people in the
meeting could hear the "hellohello", and then that's it.

Is it possible to have a auto-muted auto-pickup call?

Rob



 

Get your own web address.  
Have a HUGE year through Yahoo! Small Business.
http://smallbusiness.yahoo.com/domains/?p=BESTDEAL
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Re: [asterisk-users] Automatic Dial, Play message

[Yuan LIU]

From: Stefan Wintermeyer <[EMAIL PROTECTED]>
Date: Thu, 8 Feb 2007 21:56:11 +0100

Am 08.02.2007 um 18:39 schrieb Forrest Beck:

Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175

So it would pull each record with phone number, dial the number, when
answered play a pre-recorded message.


Have a look at an e-mail which I send yesterday to this list. It  contains 
a simple example for a call file. That is the way you want  to go. With 
that you can create a script which solves your problem.


  Stefan


I looked this and  
http://voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message, both 
using call files.  Can the same commands be used from inside extensions.conf 
to do same?


Yuan Liu


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Re: [asterisk-users] Asterisk outbound calling does not wait for answer before playback

[Robert DeVries]

Although probably not what you want to hear, I don't think there is any way
that Asterisk will detect answering supervision on an analog POTS line (I
believe that there used to be an option with some Telcos to get a polarity
reversal upon answering, but I don't think that is possible any more.)  If
you were willing to make the calls using a VOIP provider like Voipjet you
would not have this problem, but that might not meet your needs for other
reasons.

On 2/8/07, Alvin Austin <[EMAIL PROTECTED]> wrote:


Hello Asteriskers, :-)

We're trying to set up an outbound notification calling for system
alerts with Asterisk 1.4.0.  We generate a call file in
/var/spool/asterisk/outgoing and the outbound call is originated through
Zap/1 (Sangoma A200D to a Canadian POTS line).  The problem is that
Asterisk does not wait for the other side to answer before it starts
playing the message.  So the person called answers the phone after the
second or third ring and only hears the tail end of the message and the
"goodbye".

Ideally, we want to deliver the message immediately after the person
answers, or if an answering machine picks up, right after the "beep".

Any suggestions?

(1) The call file generator script (works ok):
#!/bin/sh

TMPFILE=`mktemp /tmp/tmp.XXX` || exit 1
echo "TMPFILE = $TMPFILE"

cat < $TMPFILE
Channel: Zap/g1/phone_number_here
Callerid: SYSTEM
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: dialout
Extension: s
Priority: 1
EOT

mv -v $TMPFILE /var/spool/asterisk/outgoing

(2) The dialout context in extensions.conf (problem - starts playback
before call is answered)
[dialout]
exten => s,1,NoOp(Dialout)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=8)
exten => s,n,Set(MACHINE=0)
exten => s,n,Answer
exten => s,n,BackgroundDetect(silence/5,1000,50)
exten => s,n,NoOp(Ans Machine detected)
exten => s,n,Set(MACHINE=1)
exten => s,n,BackgroundDetect(silence/30,1000,50,30050)
exten => s,n,NoOp(Ans Machine Message Too Long)
exten => s,n,Hangup

exten => talk,1,GotoIf($[${MACHINE}=1]?machine:human)
exten => talk,2(machine),Goto(dialout-machine,s,1)
exten => talk,3(human),Goto(dialout-human,s,1)

[dialout-machine]
exten => s,1,NoOp(Dialout to Ans Machine)
exten => s,n,Playback(/tmp/asterisk-recording)
exten => s,n,Wait(1)
; we'd like to do something to wait for the "beep" here...
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup

[dialout-human]
exten => s,1,NoOp(Dialout to Human)
exten => s,n,Playback(/tmp/asterisk-recording)
exten => s,n,Wait(1)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup


(3) *CLI>
-- Attempting call on Zap/1/1234567 for [EMAIL PROTECTED]:1 (Retry 1)
   > Channel Zap/1-1 was answered.
-- Executing [EMAIL PROTECTED]:1] NoOp("Zap/1-1", "Dialout") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("Zap/1-1", "TIMEOUT(digit)=5") in new
stack
-- Digit timeout set to 5
-- Executing [EMAIL PROTECTED]:3] Set("Zap/1-1", "TIMEOUT(response)=8") in
new stack
-- Response timeout set to 8
-- Executing [EMAIL PROTECTED]:4] Set("Zap/1-1", "MACHINE=0") in new stack
-- Executing [EMAIL PROTECTED]:5] Answer("Zap/1-1", "") in new stack
(Problem: Asterisk does not wait until the call is answered on the far
end!)
-- Executing [EMAIL PROTECTED]:6] BackgroundDetect("Zap/1-1",
"silence/5|1000|50") in new stack
-- Playing 'silence/5' (language 'en')
-- Executing [EMAIL PROTECTED]:1] GotoIf("Zap/1-1", "0?machine:human")
in new stack
-- Goto (dialout,talk,3)
-- Executing [EMAIL PROTECTED]:3] Goto("Zap/1-1", "dialout-human|s|1")
in new stack
-- Goto (dialout-human,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp("Zap/1-1", "Dialout to Human")
in new stack
-- Executing [EMAIL PROTECTED]:2]
Playback("Zap/1-1","/tmp/asterisk-recording") in new stack
-- Playing '/tmp/asterisk-recording' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Wait("Zap/1-1", "1") in new stack
-- Executing [EMAIL PROTECTED]:4] Playback("Zap/1-1", "vm-goodbye")
in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing [EMAIL PROTECTED]:5] Hangup("Zap/1-1", "") in new stack
  == Spawn extension (dialout-human, s, 5) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
[Feb  8 13:29:37] NOTICE[32512]: pbx_spool.c:351 attempt_thread: Call
completed to Zap/1/1234567

Thanks for any ideas on this!

Alvin

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Re: [asterisk-users] Cisco 7960 TFTP Timeout Error on RINGLIST.DAT and dialplan.xml

[Steve Edwards]

On Thu, 2007-02-08 at 13:27 -0500, Brian M. Arlinghaus wrote:

I've looked around and couldn't find much on this, but using two different
TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT
and dialplan.xml files.  On both the TFTP servers and the phone, I get TFTP
Timeout Errors.

The SIP configuration files load fine.

Any ideas?


Take the phone out of the equation.

Make sure iptables isn't getting in the way -- not likely since you can 
get the SIP files.


sudo /etc/init.d/iptables stop

Set lots'o "-v's" in the server_args in /etc/xinet.d/tftp and try it from 
the tftp server's command line:


tftp  -c get dialplan.xml

and hope that tftp says something useful or tftpd logs something useful in 
the sytem error log.


Move on to another host and repeat the tftp command.

Try it again from the phone.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Recording and MWI

[Michael Winstead]

Greetings List,
I am a newbie and first time mailer so bear with me. I have 2
questions.

1. recording: I have an Meridian Option 11 hooked to my Asterisk box via a
PRI with QSIG signalling. I have set up an access code of "8" in the option
11 to access the PRi to the Asterisk Box. Is there a way to set up the
record application so that a user could dial "8" and then the number based
on the caller id of the user? That is to say if I wanted ext 4711 to be
recorded each time it dialed through the asterisk box no matter what number
was dialed, how would I set that up?

2. MWI: Has anyone had any experience, or is it possible to send a Message
Waiting Indication to an Option 11 over the "D" channel on the PRI?

Thanks for any input you can provide!

Michael
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RE: [asterisk-users] Diagnosing poor call quality

[Chris Bagnall]

> Beware that ADSL uses vastly more bandwidth than you expect on small
> packets, eg if you are classifying using a cheap router then you
> probably need to at least half your claimed bandwidth in order to
> make the prioritisation work correctly.  I added some (hack) patches
> to fix the linux calculation for HTB on the linux QOS list a year or
> two back. If you have a linux router you could use those to improve
> the calculation quality for QOS - or else I found a Draytek router
> does impressively well at getting it right for small sites...  

Each site is using an old PIII-era PC running m0n0wall (www.m0n0.ch/wall) as
a router. Network cards are all decent-quality Intel Pro/100+ cards.

I've not had any complaints from either site today - I made the RTP change
on the SPA942s yesterday, and last night changed some of the QoS settings on
m0n0wall at the ip290 site.

> Very likely you will find that the issue is variable jitter on the
> line.  The link above should help you figure this out 

If this is the case, would upgrading to 1.4 with the new SIP jitter buffer
help at all?

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [asterisk-users] error when compiling zaptel-1.4

[Yuan LIU]

From: Richard Lyman <[EMAIL PROTECTED]>
Date: Thu, 08 Feb 2007 13:21:58 -0800


when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory.  Stop.


Seems to say you don't have full kernel source.  That's a requirement for 
kernel 2.4.


Yuan Liu

...
i can't believe noone has mentioned he did a 'make linux26' when his kernel 
is obviously a 2.4


Can't believe myself:-)

Yuan Liu


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RE: [asterisk-users] Best phone for easy provisioning

[shadowym]

Have you tried them lately???
 
Firmware and documentation is light years ahead of where it was a year ago!
NONE of your issues are issues any more IMHO.

  _  

From: Michelle Dupuis [mailto:[EMAIL PROTECTED] 
Sent: Thursday, February 08, 2007 10:00 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Best phone for easy provisioning


We used Aastra's for a good while, but gave up on them (and switched to
Cisco).   Aastra's seem cheaper up front (hardware costs), but the time
wasted chasing firmware bugs, lack of documentation, and poor support
quickly eat up any savings.  (unless your needs are very basic).
 
MD

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Thursday, February 08, 2007 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best phone for easy provisioning


I liked polycom a lot.

- Original Message - 
From: Rod Bacon   
To: asterisk-users@lists.digium.com 
Sent: Thursday, February 08, 2007 10:45 AM
Subject: [asterisk-users] Best phone for easy provisioning


Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I've used CISCO 79XX phones, and
they're great (but too expensive). I like Grandstream phones, but their
provisioning sucks. 

 

What is everybody else using in large environments where individual config
is not an option?

 



Rod Bacon

Technical Manager

JASCO Consulting Pty. Ltd.

  http://www.jasco.net.au

Ph. 03 9432 6376

Fax: 03 9432 6378



 



  _  




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Re: [asterisk-users] Asterisk 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)

[Anthony Kepler]

Did you ever find a solution for this?  I'm in the same boat with 
1.4.0-beta3 and SpanDSP


   - Anthony Kepler

Matt Gibson wrote:

Okay, So,

More updates after testing some more

1. with the "free" line commented out of app_rxfax.c, and recompiled,
asterisk seems to work on non-fax incoming calls to my fax extension.
Doesn't send a file obviously, but does seem to actually reach the
right place and do what it's supposed to do.

2. Here is all the debug output I have while trying to get this to
receive the faxes:

Error #1 :
--
*** glibc detected *** asterisk: double free or corruption (out): 
0x0820ae98 ***

=== Backtrace: =
/lib/libc.so.6[0xb72e8d20]
/lib/libc.so.6(__libc_free+0x84)[0xb72ea364]
/lib/libc.so.6(closedir+0x28)[0xb730bf48]
/usr/lib/asterisk/modules/app_voicemail.so[0xb66d1ef5]
/usr/lib/asterisk/modules/app_voicemail.so[0xb66d26a4]
/usr/lib/asterisk/modules/chan_zap.so[0xb697304e]
asterisk[0x80ecb99]
/lib/libpthread.so.0[0xb7e6b294]
/lib/libc.so.6(__clone+0x5e)[0xb7341c9e]
=== Memory map: 
08048000-0813b000 r-xp  03:03 2474075/usr/sbin/asterisk
0813b000-08148000 rw-p 000f3000 03:03 2474075/usr/sbin/asterisk
08148000-0822b000 rw-p 08148000 00:00 0  [heap]
b5e0-b5e21000 rw-p b5e0 00:00 0
b5e21000-b5f0 ---p b5e21000 00:00 0
b5fc1000-b5fc2000 ---p b5fc1000 00:00 0
b5fc2000-b5ffd000 rwxp b5fc2000 00:00 0
b602e000-b602f000 ---p b602e000 00:00 0
b602f000-b606a000 rwxp b602f000 00:00 0
b606a000-b609b000 r-xp  03:03 889758 
/usr/lib/libcurl.so.3.0.0
b609b000-b609c000 rw-p 00031000 03:03 889758 
/usr/lib/libcurl.so.3.0.0

b60a4000-b60a8000 r-xp  03:03 1066578/usr/lib/libogg.so.0.5.2
b60a8000-b60a9000 rw-p 3000 03:03 1066578/usr/lib/libogg.so.0.5.2
b60a9000-b60b4000 r-xp  03:03 1130913
/usr/lib/libvorbisenc.so.2.0.2
b60b4000-b61a3000 rw-p a000 03:03 1130913
/usr/lib/libvorbisenc.so.2.0.2

b61a3000-b61a5000 rw-p b61a3000 00:00 0
b61a5000-b61bf000 r-xp  03:03 1132171
/usr/lib/libvorbis.so.0.3.1
b61bf000-b61cd000 rw-p 0001a000 03:03 1132171
/usr/lib/libvorbis.so.0.3.1

b61d2000-b61d4000 r-xp  03:03 2376646
/usr/lib/asterisk/modules/func_curl.so
b61d4000-b61d5000 rw-p 1000 03:03 2376646
/usr/lib/asterisk/modules/func_curl.so
b61d5000-b61d6000 ---p b61d5000 00:00 0
b61d6000-b6211000 rwxp b61d6000 00:00 0
b6211000-b6212000 ---p b6211000 00:00 0
b6212000-b6252000 rwxp b6212000 00:00 0
b6252000-b6253000 ---p b6252000 00:00 0
b6253000-b6293000 rwxp b6253000 00:00 0
b6293000-b629b000 r-xp  03:03 1002454/lib/libnss_nis-2.4.so
b629b000-b629d000 rw-p 7000 03:03 1002454/lib/libnss_nis-2.4.so
b629d000-b62a3000 r-xp  03:03 1002470
/lib/libnss_compat-2.4.so
b62a3000-b62a5000 rw-p 5000 03:03 1002470
/lib/libnss_compat-2.4.so

b62a9000-b62ac000 r-xp  03:03 2376631
/usr/lib/asterisk/modules/format_ogg_vorbis.so
b62ac000-b62ad000 rw-p 2000 03:03 2376631
/usr/lib/asterisk/modules/format_ogg_vorbis.so
b62ad000-b62b r-xp  03:03 1343898
/usr/lib/pwlib/codecs/audio/g726_audio_pwplugin.so
b62b-b62b1000 rw-p 3000 03:03 1343898
/usr/lib/pwlib/codecs/audio/g726_audio_pwplugin.so
b62b1000-b62b9000 r-xp  03:03 1343897
/usr/lib/pwlib/codecs/audio/lpc10_audio_pwplugin.so
b62b9000-b62ba000 rw-p 8000 03:03 1343897
/usr/lib/pwlib/codecs/audio/lpc10_audio_pwplugin.so
b62ba000-b62c7000 r-xp  03:03 1343896
/usr/lib/pwlib/codecs/audio/ilbc_audio_pwplugin.so
b62c7000-b62ca000 rw-p c000 03:03 1343896
/usr/lib/pwlib/codecs/audio/ilbc_audio_pwplugin.so
b62ca000-b62e4000 r-xp  03:03 1343895
/usr/lib/pwlib/codecs/audio/speex_audio_pwplugin.so
b62e4000-b62e9000 rw-p 0001a000 03:03 1343895
/usr/lib/pwlib/codecs/audio/speex_audio_pwplugin.so
b62e9000-b62f2000 r-xp  03:03 1343907
/usr/lib/pwlib/codecs/audio/gsm0610_audio_pwplugin.so
b62f2000-b62f3000 rw-p 8000 03:03 1343907
/usr/lib/pwlib/codecs/audio/gsm0610_audio_pwplugin.so
b62f3000-b633d000 r-xp  03:03 2376522
/usr/lib/asterisk/modules/chan_h323.so
b633d000-b6341000 rw-p 00049000 03:03 2376522
/usr/lib/asterisk/modules/chan_h323.so
b6341000-b639b000 r-xp  03:03 3244836
/opt/swift/lib/libceplex_us.so.4.1
b639b000-b639e000 rw-p 00059000 03:03 3244836
/opt/swift/lib/libceplex_us.so.4.1
b639e000-b63ba000 r-xp  03:03 3244833
/opt/swift/lib/libceplang_en.so.4.1
b63ba000-b63c rw-p 0001b000 03:03 3244833
/opt/swift/lib/libceplang_en.so.4.1
b63c-b643c000 r-xp  03:03 3244839
/opt/swift/lib/libswift.so.4.1
b643c000-b6443000 rw-p 0007b000 03:03 3244839
/opt/swift/lib/libswift.so.4.1

b6443000-b6446000 r-xp  03:03 2376595
/usr/lib/asterisk/modules/app_txfax.so
b6446000-b6447000 rw-p 2000 03:03 2376595
/usr/lib/asterisk/modules/app_txfax.so
b6447000-b644a000 r-xp  03:03 2376580
/usr/lib/asterisk/modules/app_rxfax.so
b644a000-b644b000 rw-p 2000 03:03 2376580
/usr/lib/asteri

Re: [asterisk-users] Asterisk and 802.11g

[Yuan LIU]

From: Jason Fuermann <[EMAIL PROTECTED]>
Date: Thu, 08 Feb 2007 10:33:26 -0600

your asterisk box has to do audio conversion, its getting bogged down


I realize that I forgot to give some important details.  Actually the VoIP 
caller in question is from another Asterisk sitting on the same LAN but on 
Ethernet.  Additionally, calls from a SIP hard phone (GrandStream, pictured 
as VoIP extension in the original drawing from same LAN) directly to the 
Wi-Fi Asterisk into PSTN have not caused any network blockage.  So the 
complete scenarios are like this:


1. <--- FXS-Asterisk-Asterisk-FXO,

caller
  |_ FXS _ *A --- 802.11g --- *B
   |
 FXO ___ PSTN ___ recepient

---> bad, causing Wi-Fi blockage

2. <--- FXS-Asterisk-Asterisk-Console,

caller
  |_ FXS _ *A --- 802.11g --- *B
   |
  Console ___ recepient

---> acceptable, no Wi-Fi congestion

3. <--- GrandStream-Asterisk-PSTN,

caller
  |_ GrandStream --- 802.11g --- *B
|
  FXO ___ PSTN ___ recepient

---> acceptable, no Wi-Fi congestion

Each scenario engages the same Wi-Fi network to the same Asterisk box (*B).  
The network blockage only occurs when two Asterisk boxes are involved. 
(Senario 1)


Any idea? (There's little extra traffic in Wi-Fi, nor is there any other 
active channel in any Asterisk.)


Yuan Liu


Thanks for your reply, Jason.  Two further questions:
1) I thought all networking would be done in the card, not taxing CPU much?
2) I get reasonable quality (and no significant network blockage) when I 
answer call from Console.  In this case, Asterisk also needs to "transcode" 
audio into the sound card, right?  G.711 is supposed to be the least taxing 
CODEC, and I'm pretty sure I'm using G.711 between VoIP extensions (hence 
no transcoding between VoIP and FXO. (disallow => all, allow = ulaw)


Yuan Liu


Yuan LIU wrote:
I'm greatly surprised when testing an Asterisk box with 802.11g.  Here's 
the topology:


VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
  |
FXO ___ PSTN extension

When I call a VoIP extension on that box (from a VoIP extension), voice 
is good.  But when this box tries to bridge the call with a PSTN 
extension, voice is completely broken.  And it's not because of the cheap 
X100P - when I ping the box, round trip is >4,000 ms, most of the time 
causing timeout.  Once the call hangs up, ping time dropped to 1-2 ms.  
Ping time started to surge even when FXO is simply ringing.


If VoIP to VoIP extension call uses re-invite (which it did), voice is 
also good in the Console channel.


How can voice traffic stall 802.11g? (I haven't checked, but CODEC is 
likely ulaw.)



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[asterisk-users] Re: Disconnect Supervision UK / BT solution?

[Chris Earle]

Wondering if you ever got this change made and if it did anything?

Update us if you please :-)

--
Chris



"Chris Earle (CBL)" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> thanks for your helpful investigation!  I await news :-)
>
> --
> Chris
>
>
> - Original Message - 
> From: "Matt Brown" <[EMAIL PROTECTED]>
> To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing
List -
> Non-Commercial Discussion" 
> Sent: Saturday, January 20, 2007 7:55 AM
> Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution?
>
>
> > Well,
> >
> > I have just phoned BT today who said they can increase the CPC value
> > on the line - however it needs to be done at the exchange - and has
> > been booked for Tues.
> >
> > I suppose I will know wether this worked on Tues :-) - I shall post
> > my findings.
> >
> > Regards
> >
> > --
> > Matt Brown
> >
> >
> >
> > On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote:
> >
> > > Hi all
> > >
> > > I'm using sangoma a200 cards in the UK and have the ongoing, often
> > > noted
> > > problem of disconnect supervision with BT POTS lines.
> > >
> > > Just noticed this post on
> > > http://www.voip-info.org/wiki/view/UK+Asterisk+Details
> > > stating that potentially someone's got a solution :
> > >
> > > "TDM400P & Not Detecting Hangups:
> > >
> > >  Got a TDM400P installed and having problems with Asterisk not
> > > detecting
> > > hangups? Using BT? If so, contact BT and ask what the "Disconnect
> > > Clear
> > > Time" setting is for your phone line. Odds are it's probably 100.
> > > Increasing
> > > it to 800 fixed the issue for me.
> > >
> > > "Disconnect Clear Time" is BT's name for CPC. "
> > >
> > >
> > > Does anyone have any thoughts/confirmation about this finally being
> > > a viable
> > > solution?  This disconnect supervision problem has plagued TDM and
> > > Sangoma
> > > cards for a long time!
> > >
> > > Comments appreciated before I get on the phone with BT
> > >
> > >
> > > --
> > > Chris Earle
> > > System Solutions Specialist
> > >
> > >
> > > -- 
> > > This message has been scanned for viruses and
> > > dangerous content and is believed to be clean.
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> >
> > -- 
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> > dangerous content and is believed to be clean.
>
>
> -- 
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Re: [asterisk-users] Automatic Dial, Play message

[Matt Florell]

On 2/8/07, Forrest Beck <[EMAIL PROTECTED]> wrote:

Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175

So it would pull each record with phone number, dial the number, when
answered play a pre-recorded message.

It could be used to notify parents at a school that a after school
game is canceled.

I appreciate any direction you can point me in.


I'm sure it's more than you are looking for, but VICIDIAL can do what
you want to do. It's GPL and is in use doing auto-messaging campaigns
right now for several businesses and governmental organizations:
http://astguiclient.sourceforge.net/vicidial.html

MATT---
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Re: [asterisk-users] Best phone for easy provisioning

[Bryan M. Johns]

I can only speak for Aastra phones.

Central provisioning is very easy.  All you need is one simple text  
file on a TFTP, FTP, or HTTP server which all the phones point to.   
To customize individual phones you add a second text file for each  
phone you want customized.  The custom text file is given the name  
of the phones MAC address.  When the phone reboots it first reads  
the general text file and then reads it's custom file which will  
overwrite any duplicate setting in the general text file.


To remotely reconfigure and reboot phones you can configure them to  
check for updates to these files or for updated firmware at a  
certain time of day.  You can also remotely reboot individual  
phones based on extension from the Asterisk CLI.  Of course, you  
can also access an individual phones Webpage configuration based on  
it's IP address.


From: Rod Bacon [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 08, 2007 12:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Best phone for easy provisioning

Does anyone have any recommendations for a phone that has easy to  
understand/implement central provisioning? I’ve used CISCO 79XX  
phones, and they’re great (but too expensive). I like Grandstream  
phones, but their provisioning sucks.




What is everybody else using in large environments where individual  
config is not an option?






Rod Bacon

Technical Manager

JASCO Consulting Pty. Ltd.

http://www.jasco.net.au

Ph. 03 9432 6376

Fax: 03 9432 6378



Polycom's central provisioning is very straight forward and very  
powerful.


There is support for all major connectivity methods (tftp, ftp, ftps,  
http, https, etc) and the configuration capability is more broad than  
any other phone we have worked with.


I hope this information is helpful.

Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com


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Re: [asterisk-users] Disconnection supervision: what about PBX

[Eric \"ManxPower\" Wieling]

Tzafrir Cohen wrote:

On Thu, Feb 08, 2007 at 01:38:30PM -0500, C F wrote:

This device can solve many problems, and is a must for most
applications where asterisk is connected using FXO ports and the host
PBX deosn't give CPC.
http://www.sandman.com/wizard.html#CPCGenerator


How does it compare to busydetect of chan_zap ?



Maybe it works well.
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[asterisk-users] Auto Answer (Paging)

[Rob Schall]

I'm trying to duplicate a behavior we had with our old avaya system, and
I've come across Auto Answer (Ring Answer). However, its not quite the
same yet.

Right now, when I dial **5053, it will add the SIP header for Ring
Answer and it will call 5053. The phone auto pickups just fine. However,
we need that call to be muted. If you were to call into a meeting, we
wouldn't want them to hear that meeting, but instead the people in the
meeting could hear the "hellohello", and then that's it.

Is it possible to have a auto-muted auto-pickup call?

Rob

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Re: [asterisk-users] does any one knows of a Softphone that works under terminal services?

[Manrique Feoli]

I wonder if I setup a softphone on each terminal if they will actually 
work as independent phones well enough, but haven´t tested it.


MF escribió:

Hi all

I'm looking for a softphone that works well under terminal services 
environment,


we need to set up  24 to 32 phones for a call center,

also, does any one knows if it  will actually work fine under load?


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[asterisk-users] www.BarCampUSA.org tickets went on sale this week

[Dean Collins]

Hi Guys and Girls, Freaks and Geeks,

 

I know you all had a blast at least years Astricon and are looking
forward to this years as well.however that's not why I'm writing
to you today. I know most of you are familiar with the www.Barcamp.org
  events I'm writing to let you know about a
'mega' barcamp that is occurring on the 23rd to 26th of August 2007
www.BarCampUSA.org   

 

 

 

 

What would you want to do to the BarCamp concept if you could change
some things...

 

 

Q.  "How about make it longer" A.
Well naturally. How would 24 hours a day for 4 days straight suit
you? 

Q.  "How about having more people"...   A.   I was
thinking about the same thing, would 5,000+ people be enough for you?

Q.  "How about having entertainment at night"..  A.   Would a
couple of different bands, video DJ's, and film demonstrations fit that
requirement (oh and a LAN Party to die for).

Q   "Can I be involved"  A.
Yep discussion topics, volunteering committees, organization committees
are being formed now.



 

 

If this sounds like something of interest to you then head over to
www.BarCampUSA.org   

It's in Wisconsin  (yeh I knowbut I hear they don't bite...:-) and
they have electricity and the internet in Wisconsin as well..oh and
they have a space big enough for 5,000-10,000+ of us to get together -
want to try organize that in New York?)

 

If you want to help out the Asterisk cause then go and register at
http://www.barcampusa.com/wiki/index.php?title=Asterisk  and include
details on what you want your topic to be about.

 

 

Discounts and benefits for people who buy tickets or companies that
sponsor before 1st of May.

 

Tickets went on sale this week at http://www.barcampusa.com/tickets.htm 

 

If you work for a company that might like to help keep costs to a
minimum by being a sponsor then check out
http://www.barcampusa.com/wiki/index.php?title=Sponsors 

 

(oh and btw if you are part of a startup looking for fundingcheck
out an event I'm organizing
http://www.barcampusa.com/wiki/index.php?title=SpeedPitchShootout )

 

Please also forward this email to anyone who you know would be
interested, especially if they live overseas, I'm really trying to make
this as global event as possible. People are already calling this the
"Woodstock of our generation"  so lets try and make it as special as
this opportunities deserves.

 

 

Regards,

 

Dean Collins

P.S. this will be the last email I send to the list about BarCamp until
at least August so if you read this far and think I wasted your
bandwidth chill until then ok :-)

 

 

 

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Re: [asterisk-users] Cisco 7960 TFTP Timeout Error on RINGLIST.DAT and dialplan.xml

[Patrick]

On Thu, 2007-02-08 at 13:27 -0500, Brian M. Arlinghaus wrote:
> I've looked around and couldn't find much on this, but using two different 
> TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT 
> and dialplan.xml files.  On both the TFTP servers and the phone, I get TFTP 
> Timeout Errors.
> 
> The SIP configuration files load fine.
> 
> Any ideas?

Have you made sure that the file has the proper rights? Iirc it needs to
be 644. You can also use Wireshark (former Ethereal) to sniff the
traffic and see what the Cisco requests.

Regards,
Patrick

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Re: [asterisk-users] error when compiling zaptel-1.4

[Rodrigo Gonzalez]

Alyed Tzompa wrote:

The error lies here:

 >make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
 >make: *** arch/i386/boot: No such file or directory. Stop.

do you have the kernel-headers installed? (e.g. 
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora)



Alyed


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when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory. Stop.
make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2


You have a debian there I think cause of the kernel name and version.

2 things

uname -r

Make sure that the running kernel is the same version 2.4.27-3-386

Second run

make

To use make linux26 you need to have kernel 2.6

You can install kernel 2.6, kernel-headers for the 2.6 version and the 
you can use make linux26




any idea
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Re: [asterisk-users] error when compiling zaptel-1.4

[Tzafrir Cohen]

On Thu, Feb 08, 2007 at 11:55:24AM -0800, Yuan LIU wrote:
> >From: "younss azzayani" <[EMAIL PROTECTED]>
> >Date: Thu, 8 Feb 2007 17:58:08 +
> >
> >when i compile zaptel
> >make linux26

With 1.4: just 'make'

> >make install
> >i got these errors:
> >
> >make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
> >make -C datamods clean
> >make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
> >make -C /lib/modules/2.4.27-3-386/build

Is this link something you have added yourself, or something that "was
there"?

> >SUBDIRS=/usr/src/zaptel-1.4/datamods clean
> >make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
> >make: *** arch/i386/boot: No such file or directory.  Stop.
> 
> Seems to say you don't have full kernel source.  That's a requirement for 
> kernel 2.4.

kernel-headers should do:

This seems to be a Debian Sarge system:

  apt-get install kernel-headers-`uname -r`

Though I tend to believe you already have it.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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re: [asterisk-users] Asterisk outbound calling does not wait for answer before playback

[Alyed Tzompa]

Had the same issue time ago, but Eric shed good light on it, 
have a look at:

http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html

Summary: sorry, no nice work around.

Alyed  


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Hello Asteriskers, :-)

We're trying to set up an outbound notification calling for system 
alerts with Asterisk 1.4.0.  We generate a call file in 
/var/spool/asterisk/outgoing and the outbound call is originated through 
Zap/1 (Sangoma A200D to a Canadian POTS line).  The problem is that 
Asterisk does not wait for the other side to answer before it starts 
playing the message.  So the person called answers the phone after the 
second or third ring and only hears the tail end of the message and the 
"goodbye".

Ideally, we want to deliver the message immediately after the person 
answers, or if an answering machine picks up, right after the "beep".

Any suggestions?

(1) The call file generator script (works ok):
#!/bin/sh

TMPFILE=`mktemp /tmp/tmp.XXX` || exit 1
echo "TMPFILE = $TMPFILE"

cat < $TMPFILE
Channel: Zap/g1/phone_number_here
Callerid: SYSTEM
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: dialout
Extension: s
Priority: 1
EOT

mv -v $TMPFILE /var/spool/asterisk/outgoing

(2) The dialout context in extensions.conf (problem - starts playback 
before call is answered)
[dialout]
exten => s,1,NoOp(Dialout)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=8)
exten => s,n,Set(MACHINE=0)
exten => s,n,Answer
exten => s,n,BackgroundDetect(silence/5,1000,50)
exten => s,n,NoOp(Ans Machine detected)
exten => s,n,Set(MACHINE=1)
exten => s,n,BackgroundDetect(silence/30,1000,50,30050)
exten => s,n,NoOp(Ans Machine Message Too Long)
exten => s,n,Hangup

exten => talk,1,GotoIf($[${MACHINE}=1]?machine:human)
exten => talk,2(machine),Goto(dialout-machine,s,1)
exten => talk,3(human),Goto(dialout-human,s,1)

[dialout-machine]
exten => s,1,NoOp(Dialout to Ans Machine)
exten => s,n,Playback(/tmp/asterisk-recording)
exten => s,n,Wait(1)
; we'd like to do something to wait for the "beep" here...
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup

[dialout-human]
exten => s,1,NoOp(Dialout to Human)
exten => s,n,Playback(/tmp/asterisk-recording)
exten => s,n,Wait(1)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup

(3) *CLI>
-- Attempting call on Zap/1/1234567 for [EMAIL PROTECTED]:1 (Retry 1)
   > Channel Zap/1-1 was answered.
-- Executing [EMAIL PROTECTED]:1] NoOp("Zap/1-1", "Dialout") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("Zap/1-1", "TIMEOUT(digit)=5") in new 
stack
-- Digit timeout set to 5
-- Executing [EMAIL PROTECTED]:3] Set("Zap/1-1", "TIMEOUT(response)=8") in 
new stack
-- Response timeout set to 8
-- Executing [EMAIL PROTECTED]:4] Set("Zap/1-1", "MACHINE=0") in new stack
-- Executing [EMAIL PROTECTED]:5] Answer("Zap/1-1", "") in new stack
(Problem: Asterisk does not wait until the call is answered on the far end!)
-- Executing [EMAIL PROTECTED]:6] BackgroundDetect("Zap/1-1", 
"silence/5|1000|50") in new stack
-- Playing 'silence/5' (language 'en')
-- Executing [EMAIL PROTECTED]:1] GotoIf("Zap/1-1", "0?machine:human") 
in new stack
-- Goto (dialout,talk,3)
-- Executing [EMAIL PROTECTED]:3] Goto("Zap/1-1", "dialout-human|s|1") 
in new stack
-- Goto (dialout-human,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp("Zap/1-1", "Dialout to Human") 
in new stack
-- Executing [EMAIL PROTECTED]:2] 
Playback("Zap/1-1","/tmp/asterisk-recording") in new stack
-- Playing '/tmp/asterisk-recording' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Wait("Zap/1-1", "1") in new stack
-- Executing [EMAIL PROTECTED]:4] Playback("Zap/1-1", "vm-goodbye") 
in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing [EMAIL PROTECTED]:5] Hangup("Zap/1-1", "") in new stack
  == Spawn extension (dialout-human, s, 5) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
[Feb  8 13:29:37] NOTICE[32512]: pbx_spool.c:351 attempt_thread: Call 
completed to Zap/1/1234567

Thanks for any ideas on this!

Alvin

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Re: [asterisk-users] error when compiling zaptel-1.4

[Richard Lyman]

Yuan LIU wrote:

From: "younss azzayani" <[EMAIL PROTECTED]>
Date: Thu, 8 Feb 2007 17:58:08 +

when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory.  Stop.


Seems to say you don't have full kernel source.  That's a requirement 
for kernel 2.4.


Yuan Liu


make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2

any idea


i can't believe noone has mentioned he did a 'make linux26' when his 
kernel is obviously a 2.4




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Re: [asterisk-users] Disconnection supervision: what about PBX

[Tzafrir Cohen]

On Thu, Feb 08, 2007 at 01:38:30PM -0500, C F wrote:
> This device can solve many problems, and is a must for most
> applications where asterisk is connected using FXO ports and the host
> PBX deosn't give CPC.
> http://www.sandman.com/wizard.html#CPCGenerator

How does it compare to busydetect of chan_zap ?

-- 
   Tzafrir Cohen   
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Any Way to Get # Functionality in DISA

[Robert DeVries]

When using a SIP phone with Asterisk, hitting the # key (pound or hash
depending on where in the world you happen to be) tells Asterisk that there
are no more digits coming, and to put the call through immediately based on
the digits already entered.  This is the same functionality as the PSTN (at
least in North America).

However, DISA just sees the # as another digit, and therefore pressing #
produces an error.   I suppose it would be possible to write something in
the dial plan that looks for a # then strips the # and puts the call
through, but I'm curious if there is any other way to do this.  For one
thing, this would require having a special context for DISA, rather than
just having DISA place calls through an existing context.
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Re: [asterisk-users] Automatic Dial, Play message

[Stefan Wintermeyer]

Am 08.02.2007 um 18:39 schrieb Forrest Beck:

Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175

So it would pull each record with phone number, dial the number, when
answered play a pre-recorded message.


Have a look at an e-mail which I send yesterday to this list. It  
contains a simple example for a call file. That is the way you want  
to go. With that you can create a script which solves your problem.


  Stefan

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998


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Re: [asterisk-users] Automatic Dial, Play message

[Lee Jenkins]

Forrest Beck wrote:

Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175



I'm currently working on an AGI/GUI application to do this right now as 
a part of another, larger project.  It uses a FirebirdSQL database to 
store lists of people to broadcast messages to.  There will be a 
Windows/Linux based GUI to control/configure it as well.


It should be a ready in a month or so.  Send me a private email if you 
don't find anything by then.



--

Warm Regards,

Lee

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re: [asterisk-users] Automatic Dial, Play message

[Alyed Tzompa]

I've made a very simple one time ago I
could share with you, it's made on bash, takes as input a CSV file,
places the calls using the /var/spool/asterisk/outbound directory, and
restricts the number of calls to a given number at a time (say 10)

I can share it with you only if you promise NOT to use it for
Telemarketing, otherwise might the mighty spirits of (place the name of
some super natural power here) cast the most terrible spell on you
forever and ever.

so, if you still want it just contact me directly :)

Alyed  


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Thu,  8 Feb 2007 10:39:31 -0700 (MST)

Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175

So it would pull each record with phone number, dial the number, when
answered play a pre-recorded message.

It could be used to notify parents at a school that a after school
game is canceled.

I appreciate any direction you can point me in.

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]

-- 
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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re: [asterisk-users] error when compiling zaptel-1.4

[Alyed Tzompa]

The error lies here:

>make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'

>make: *** arch/i386/boot: No such file or directory.  Stop.

do you have the kernel-headers installed? (e.g.  
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora) 

Alyed 


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when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory.  Stop.
make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2

any idea
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Re: [asterisk-users] Automatic Dial, Play message

[William Piper]

This should do what you asked:
http://voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message

bp

On 2/8/07, Forrest Beck <[EMAIL PROTECTED]> wrote:


Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175

So it would pull each record with phone number, dial the number, when
answered play a pre-recorded message.

It could be used to notify parents at a school that a after school
game is canceled.

I appreciate any direction you can point me in.

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]


--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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Re: [asterisk-users] error when compiling zaptel-1.4

[Steve Kennedy]

On Thu, Feb 08, 2007 at 05:58:08PM +, younss azzayani wrote:

> when i compile zaptel
> make linux26
> make install
> i got these errors:
> make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
> make -C datamods clean
> make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
> make -C /lib/modules/2.4.27-3-386/build
> SUBDIRS=/usr/src/zaptel-1.4/datamods clean
> make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
> make: *** arch/i386/boot: No such file or directory.  Stop.
> make: Entering an unknown directorymake: Leaving an unknown
> directorymake[2]: *** [archclean] Error 2
> make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
> make[1]: *** [clean] Error 2
> make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
> make: *** [clean] Error 2
> any idea

You've got kernel 2.4 headers so either you haven't got the right
headers for the kernel or you really have a 2.4 kernel.

In which case just do
make

instead of make linux26


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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RE: [asterisk-users] Best phone for easy provisioning

[shadowym]

I can only speak for Aastra phones.
 
Central provisioning is very easy.  All you need is one simple text file on
a TFTP, FTP, or HTTP server which all the phones point to.  To customize
individual phones you add a second text file for each phone you want
customized.  The custom text file is given the name of the phones MAC
address.  When the phone reboots it first reads the general text file and
then reads it's custom file which will overwrite any duplicate setting in
the general text file.
 
To remotely reconfigure and reboot phones you can configure them to check
for updates to these files or for updated firmware at a certain time of day.
You can also remotely reboot individual phones based on extension from the
Asterisk CLI.  Of course, you can also access an individual phones Webpage
configuration based on it's IP address.

  _  

From: Rod Bacon [mailto:[EMAIL PROTECTED] 
Sent: Thursday, February 08, 2007 12:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Best phone for easy provisioning



Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I've used CISCO 79XX phones, and
they're great (but too expensive). I like Grandstream phones, but their
provisioning sucks. 

 

What is everybody else using in large environments where individual config
is not an option?

 



Rod Bacon

Technical Manager

JASCO Consulting Pty. Ltd.

  http://www.jasco.net.au

Ph. 03 9432 6376

Fax: 03 9432 6378



 

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RE: [asterisk-users] error when compiling zaptel-1.4

[Yuan LIU]

From: "younss azzayani" <[EMAIL PROTECTED]>
Date: Thu, 8 Feb 2007 17:58:08 +

when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory.  Stop.


Seems to say you don't have full kernel source.  That's a requirement for 
kernel 2.4.


Yuan Liu


make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2

any idea



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Re: [asterisk-users] Asterisk and 802.11g

[Yuan LIU]

From: "younss azzayani" <[EMAIL PROTECTED]>
Date: Thu, 8 Feb 2007 17:20:30 +

did you test to call from a soft phone using pstn, if you get a bad
sound that s mean that the zaptel param must be changed if not try to call 
from a soft phone your wirless phones and test


I've tested Zaptel with this card extensively and the voice is acceptable.  
But I'm not sure how to call from a soft phone using PSTN.  Could you 
elaborate?


Yuan Liu


2007/2/8, Yuan LIU <[EMAIL PROTECTED]>:
I'm greatly surprised when testing an Asterisk box with 802.11g.  Here's 
the topology:


VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
   |
 FXO ___ PSTN extension

When I call a VoIP extension on that box (from a VoIP extension), voice is
good.  But when this box tries to bridge the call with a PSTN extension,
voice is completely broken.  And it's not because of the cheap X100P - 
when

I ping the box, round trip is >4,000 ms, most of the time causing timeout.
Once the call hangs up, ping time dropped to 1-2 ms.  Ping time started to
surge even when FXO is simply ringing.

If VoIP to VoIP extension call uses re-invite (which it did), voice is 
also

good in the Console channel.

How can voice traffic stall 802.11g? (I haven't checked, but CODEC is 
likely ulaw.)


Yuan Liu



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[asterisk-users] Asterisk outbound calling does not wait for answer before playback

[Alvin Austin]

Hello Asteriskers, :-)

We're trying to set up an outbound notification calling for system 
alerts with Asterisk 1.4.0.  We generate a call file in 
/var/spool/asterisk/outgoing and the outbound call is originated through 
Zap/1 (Sangoma A200D to a Canadian POTS line).  The problem is that 
Asterisk does not wait for the other side to answer before it starts 
playing the message.  So the person called answers the phone after the 
second or third ring and only hears the tail end of the message and the 
"goodbye".


Ideally, we want to deliver the message immediately after the person 
answers, or if an answering machine picks up, right after the "beep".


Any suggestions?

(1) The call file generator script (works ok):
#!/bin/sh

TMPFILE=`mktemp /tmp/tmp.XXX` || exit 1
echo "TMPFILE = $TMPFILE"

cat < $TMPFILE
Channel: Zap/g1/phone_number_here
Callerid: SYSTEM
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: dialout
Extension: s
Priority: 1
EOT

mv -v $TMPFILE /var/spool/asterisk/outgoing

(2) The dialout context in extensions.conf (problem - starts playback 
before call is answered)

[dialout]
exten => s,1,NoOp(Dialout)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=8)
exten => s,n,Set(MACHINE=0)
exten => s,n,Answer
exten => s,n,BackgroundDetect(silence/5,1000,50)
exten => s,n,NoOp(Ans Machine detected)
exten => s,n,Set(MACHINE=1)
exten => s,n,BackgroundDetect(silence/30,1000,50,30050)
exten => s,n,NoOp(Ans Machine Message Too Long)
exten => s,n,Hangup

exten => talk,1,GotoIf($[${MACHINE}=1]?machine:human)
exten => talk,2(machine),Goto(dialout-machine,s,1)
exten => talk,3(human),Goto(dialout-human,s,1)

[dialout-machine]
exten => s,1,NoOp(Dialout to Ans Machine)
exten => s,n,Playback(/tmp/asterisk-recording)
exten => s,n,Wait(1)
; we'd like to do something to wait for the "beep" here...
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup

[dialout-human]
exten => s,1,NoOp(Dialout to Human)
exten => s,n,Playback(/tmp/asterisk-recording)
exten => s,n,Wait(1)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup


(3) *CLI>
   -- Attempting call on Zap/1/1234567 for [EMAIL PROTECTED]:1 (Retry 1)
  > Channel Zap/1-1 was answered.
   -- Executing [EMAIL PROTECTED]:1] NoOp("Zap/1-1", "Dialout") in new stack
   -- Executing [EMAIL PROTECTED]:2] Set("Zap/1-1", "TIMEOUT(digit)=5") in new 
stack

   -- Digit timeout set to 5
   -- Executing [EMAIL PROTECTED]:3] Set("Zap/1-1", "TIMEOUT(response)=8") in 
new stack

   -- Response timeout set to 8
   -- Executing [EMAIL PROTECTED]:4] Set("Zap/1-1", "MACHINE=0") in new stack
   -- Executing [EMAIL PROTECTED]:5] Answer("Zap/1-1", "") in new stack
(Problem: Asterisk does not wait until the call is answered on the far end!)
   -- Executing [EMAIL PROTECTED]:6] BackgroundDetect("Zap/1-1", 
"silence/5|1000|50") in new stack

   -- Playing 'silence/5' (language 'en')
   -- Executing [EMAIL PROTECTED]:1] GotoIf("Zap/1-1", "0?machine:human") 
in new stack

   -- Goto (dialout,talk,3)
   -- Executing [EMAIL PROTECTED]:3] Goto("Zap/1-1", "dialout-human|s|1") 
in new stack

   -- Goto (dialout-human,s,1)
   -- Executing [EMAIL PROTECTED]:1] NoOp("Zap/1-1", "Dialout to Human") 
in new stack
   -- Executing [EMAIL PROTECTED]:2] 
Playback("Zap/1-1","/tmp/asterisk-recording") in new stack

   -- Playing '/tmp/asterisk-recording' (language 'en')
   -- Executing [EMAIL PROTECTED]:3] Wait("Zap/1-1", "1") in new stack
   -- Executing [EMAIL PROTECTED]:4] Playback("Zap/1-1", "vm-goodbye") 
in new stack

   -- Playing 'vm-goodbye' (language 'en')
   -- Executing [EMAIL PROTECTED]:5] Hangup("Zap/1-1", "") in new stack
 == Spawn extension (dialout-human, s, 5) exited non-zero on 'Zap/1-1'
   -- Hungup 'Zap/1-1'
[Feb  8 13:29:37] NOTICE[32512]: pbx_spool.c:351 attempt_thread: Call 
completed to Zap/1/1234567


Thanks for any ideas on this!

Alvin

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Re: [asterisk-users] Billing pulses

[David Boyd]

Hi Stefano,

I have a question, how would you go about using the billing pulses to
generate an invoice/bill.  Also can you provide an ascii drawing of the
layout of the equipment as you intend to use it, they say a picture is
worth a thousand words:)


db




On Thu, 2007-02-08 at 15:13 +0100, 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] Asterisk and 802.11g

[Yuan LIU]

From: Jason Fuermann <[EMAIL PROTECTED]>
Date: Thu, 08 Feb 2007 10:33:26 -0600

your asterisk box has to do audio conversion, its getting bogged down


Thanks for your reply, Jason.  Two further questions:
1) I thought all networking would be done in the card, not taxing CPU much?
2) I get reasonable quality (and no significant network blockage) when I 
answer call from Console.  In this case, Asterisk also needs to "transcode" 
audio into the sound card, right?  G.711 is supposed to be the least taxing 
CODEC, and I'm pretty sure I'm using G.711 between VoIP extensions (hence no 
transcoding between VoIP and FXO. (disallow => all, allow = ulaw)


Yuan Liu


Yuan LIU wrote:
I'm greatly surprised when testing an Asterisk box with 802.11g.  Here's 
the topology:


VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
  |
FXO ___ PSTN extension

When I call a VoIP extension on that box (from a VoIP extension), voice is 
good.  But when this box tries to bridge the call with a PSTN extension, 
voice is completely broken.  And it's not because of the cheap X100P - 
when I ping the box, round trip is >4,000 ms, most of the time causing 
timeout.  Once the call hangs up, ping time dropped to 1-2 ms.  Ping time 
started to surge even when FXO is simply ringing.


If VoIP to VoIP extension call uses re-invite (which it did), voice is 
also good in the Console channel.


How can voice traffic stall 802.11g? (I haven't checked, but CODEC is 
likely ulaw.)


Yuan Liu



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RE: [asterisk-users] Skutch AS-66 and an X100P

[Yuan LIU]

From: "David Ruggles" <[EMAIL PROTECTED]>
Date: Thu, 8 Feb 2007 11:57:41 -0500

I finally got my X100P working and now I have a question.

I have several Skutch phone line simulators. My X100P works as expected 
with

both a POTS line and an analog PBX port, but when I use a phone line
simulator it doesn't answer the line. The phone line simulator doesn't 
power

the line until the phone set goes offhook. Asterisk shows the RED alarm and
then the alarm clearing but never detects the ring.


I don't know anything about a line simulator but your description certainly 
points to a problem with the simulator.  As I'm also doing tests on X100P, 
I'm interested to know what does a simulator give you that your PBX doesn't. 
(I wish I had a PBX to play with.)


Yuan Liu

Any suggestions?

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] Spliting video and audio

[Anselm Martin Hoffmeister]

Am Mittwoch, den 07.02.2007, 21:57 -0800 schrieb Jason Kim:
> Hi,
> 
> This is the configuration I want.
> 
> Hard Video phone<---video--->Soft Video Phone(PC)
>^
>|
>  audio
>|
>V
>Audio Only Phone
> 
> Any idea?

You could see wether having a second call that does a "ChanSpy" or
similar into your video conversation would work. With the whisper (if
you are on * 1.4) this should do duplex audio.

With some AGI it should be easy to create a call file for any calls
incoming to the PC; that in turn can initiate a call to the Audio only
phone and ChanSpy(). You should of course answer the video call first
such that the SIP channels exist at the moment of the ChanSpy function
call.
Similar thing for outgoing calls, I guess.
I did not test this though.

HTH
Anselm

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[asterisk-users] Suppliers in Canada

[asterisk]

I am looking for some Linksys and GrandStream ATAs in Canada. I am
looking for places that ship from Canada so I don't have to deal
with the clearing of customs and tax remittance.

Any suggestion?

-- 
Thanks
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Re: [asterisk-users] Disconnection supervision: what about PBX

[C F]

This device can solve many problems, and is a must for most
applications where asterisk is connected using FXO ports and the host
PBX deosn't give CPC.
http://www.sandman.com/wizard.html#CPCGenerator

On 2/6/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:

Yuan LIU wrote:
> After reading through several recent threads, I started to wonder why
> the Cisco document (and other VoIP documents) appears to present this
> issue as VoIP gateway specific.  Don't (plain old) PBX' face the same
> issue if they use analogue interfaces?  If there are analogue PBX' at
> all, how do they solve the problem?

Yes, analog PBXs have the same issues.  Don't do anything to solve the
issue.  That is way many hotels tell their guests to not let a call ring
for more than 45 seconds or the call will be billed even if it was not
answered.
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[asterisk-users] Cisco 7960 TFTP Timeout Error on RINGLIST.DAT and dialplan.xml

[Brian M. Arlinghaus]

I've looked around and couldn't find much on this, but using two different 
TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT 
and dialplan.xml files.  On both the TFTP servers and the phone, I get TFTP 
Timeout Errors.


The SIP configuration files load fine.

Any ideas?

Thanks,
Brian Arlinghaus 


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[asterisk-users] Fwd: error when compiling zaptel-1.4

[younss azzayani]

-- Forwarded message --
From: younss azzayani <[EMAIL PROTECTED]>
Date: 8 févr. 2007 17:58
Subject: error when compiling zaptel-1.4
To: Asterisk Users Mailing List - Non-Commercial Discussion



when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory.  Stop.
make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2

any idea
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Re: [asterisk-users] CD needed: no way to burn

[Bruce Reeves]

How many disk do you need? I'll burn you one and mail it to you if you want.

On 2/6/07, Tom Poe <[EMAIL PROTECTED]> wrote:


I wonder if there are CDs available for purchase.  I don't have any way
to burn one from a downloaded iso image.  Any help appreciated.
Tom
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--
Bruce
Nortex Networks
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RE: [asterisk-users] Best phone for easy provisioning

[Michelle Dupuis]

We used Aastra's for a good while, but gave up on them (and switched to
Cisco).   Aastra's seem cheaper up front (hardware costs), but the time
wasted chasing firmware bugs, lack of documentation, and poor support
quickly eat up any savings.  (unless your needs are very basic).
 
MD

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Thursday, February 08, 2007 11:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best phone for easy provisioning


I liked polycom a lot.

- Original Message - 
From: Rod Bacon   
To: asterisk-users@lists.digium.com 
Sent: Thursday, February 08, 2007 10:45 AM
Subject: [asterisk-users] Best phone for easy provisioning


Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I've used CISCO 79XX phones, and
they're great (but too expensive). I like Grandstream phones, but their
provisioning sucks. 

 

What is everybody else using in large environments where individual config
is not an option?

 



Rod Bacon

Technical Manager

JASCO Consulting Pty. Ltd.

  http://www.jasco.net.au

Ph. 03 9432 6376

Fax: 03 9432 6378



 



  _  




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[asterisk-users] error when compiling zaptel-1.4

[younss azzayani]

when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory.  Stop.
make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2

any idea
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Re: [asterisk-users] Skutch AS-66 and an X100P

[John Novack]

David Ruggles wrote:

I finally got my X100P working and now I have a question.

I have several Skutch phone line simulators. My X100P works as expected with
both a POTS line and an analog PBX port, but when I use a phone line
simulator it doesn't answer the line. The phone line simulator doesn't power the line until the phone set goes offhook. 

Toss those in the trash
Good line simulators must provide battery when idle.
Look on eBay for a TLS-2,3,4, or 5 for a good one, and frequently 
affordable.

IMO Scutch has never made anything worth a dime

John Novack
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[asterisk-users] Automatic Dial, Play message

[Forrest Beck]

Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175

So it would pull each record with phone number, dial the number, when
answered play a pre-recorded message.

It could be used to notify parents at a school that a after school
game is canceled.

I appreciate any direction you can point me in.

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]


--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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[asterisk-users] Automatic Dial, Play message

[Forrest Beck]

Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

For example, you have a database of

FirstName LastName PhoneNumber
Jon

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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Re: [asterisk-users] Best phone for easy provisioning

[Pavel Jezek]

for massive deployment phone provisioning/fw updating through web 
interface is not optimal,
best way is via config files/templates periodicaly downloaded from 
central tftp/http server...

PJ



MF wrote:
Best and easiest provisioning I´ve found imho is Snom, great web 
interfase , followed by Polycom (web interfase used to be poor and 
slow, but once you set it up, works very well)



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Re: [asterisk-users] Asterisk and 802.11g

[younss azzayani]

did you test to call from a soft phone using pstn, if you get a bad
sound that s mean that the zaptel param must be changed if not try to
call from a soft phone your wirless phones and test

2007/2/8, Yuan LIU <[EMAIL PROTECTED]>:

I'm greatly surprised when testing an Asterisk box with 802.11g.  Here's the
topology:

VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
   |
 FXO ___ PSTN extension

When I call a VoIP extension on that box (from a VoIP extension), voice is
good.  But when this box tries to bridge the call with a PSTN extension,
voice is completely broken.  And it's not because of the cheap X100P - when
I ping the box, round trip is >4,000 ms, most of the time causing timeout.
Once the call hangs up, ping time dropped to 1-2 ms.  Ping time started to
surge even when FXO is simply ringing.

If VoIP to VoIP extension call uses re-invite (which it did), voice is also
good in the Console channel.

How can voice traffic stall 802.11g? (I haven't checked, but CODEC is likely
ulaw.)

Yuan Liu


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Re: [asterisk-users] Best phone for easy provisioning

[younss azzayani]

really you can't make a difference between them, i like thomson
2007/2/8, Pavel Jezek <[EMAIL PROTECTED]>:

Chris, (or others), do you have any negative experience with Thomson
2030? it looks very promising!
I hesitate between thomson and linksys spa 922/942,
I'm not sure, what is better for bussines use :-\
snoms are probably also good, but functionality/price ratio is, imho,
better for thomson or linksys...
PJ


> Hi,
>
> I'm using a Thomason ST2030. Had difficulties in the beginning, but
> after a firmware upgrade it works fine. And autoprovisioning works good.
> Most of the parameters are described in their official (marked as
> confidential) admin documentation from their website. Even a centralized
> phonebook is possible.
>
> Chris...
>
>
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Re: [asterisk-users] Best phone for easy provisioning

[younss azzayani]

great

i join you Thomson ST is a good choice, also you can see linksys
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Re: [asterisk-users] Re: On what distribution is www.asterisknow.com basedon ?

[Vicky]

You can easily recompile asterisk with mysql logging enabled also use all
add-ons u can use on debian and any other distro ..
On 08/02/07, Chris Earle <[EMAIL PROTECTED]> wrote:


I'm tempted to rebuild my asterisk network with AsteriskNow - my
question is, can you ADD anything to it?  i.e. cdr_mysql logging?  I
thought
I saw it didn't have that

And how does it handle the hardware?  I don't use digium cards in all of
my
servers because of country issues (Junghanns in Germany, Sangoma in UK,
etc)

If it's expandable through usual package addons etc, then it would seem
there is alot of added value because of the increased EASE of
administration
over your well-maintained debian box 

Thoughts?

--
Chris


"Maxim Veksler" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
> Hello Asteriskies,
>
> Has someone tried www.asterisknow.com ?
>
> What is the package manager used? And what is the added value compared
> to the well maintained debian based asterisk ?
>
> Thanks,
>
>
> --
> Cheers,
> Maxim Veksler
>
> "Free as in Freedom" - Do u GNU ?
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Re: [asterisk-users] SIP??

[Vicky]

config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer
definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc )

On 08/02/07, Florea Igor <[EMAIL PROTECTED]> wrote:


Hi,
I'm new to *,so i apologize for stupid questions.
I'm having problem with this arhitecture:
I'm calling asterisk from behind a NAT(sjphone user) with a low band so
I'm
using GSM codec.
In extensions.conf I have:
exten => 337,1,Dial(SIP/99@)
so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2.
RTP stream between sjphone and Asterisk are ok (GSM).
The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although
ip_pbx2 sip is telling asterisk It only knows "codec 0"
Is this a config problem or a bug?
Igor,

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[asterisk-users] Skutch AS-66 and an X100P

[David Ruggles]

I finally got my X100P working and now I have a question.

I have several Skutch phone line simulators. My X100P works as expected with
both a POTS line and an analog PBX port, but when I use a phone line
simulator it doesn't answer the line. The phone line simulator doesn't power
the line until the phone set goes offhook. Asterisk shows the RED alarm and
then the alarm clearing but never detects the ring.

Any suggestions?

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] Best phone for easy provisioning

[Mike Clark]

Dovid B wrote:


I liked polycom a lot.

- Original Message -
*From:* Rod Bacon 
*To:* asterisk-users@lists.digium.com

*Sent:* Thursday, February 08, 2007 10:45 AM
*Subject:* [asterisk-users] Best phone for easy provisioning

Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I’ve used CISCO 79XX
phones, and they’re great (but too expensive). I like Grandstream
phones, but their provisioning sucks.

What is everybody else using in large environments where
individual config is not an option?


Aastra phones are pretty nice for central configuration.
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Re: [asterisk-users] Best phone for easy provisioning

[MF]

Best and easiest provisioning I´ve found imho is Snom, great web 
interfase , followed by Polycom (web interfase used to be poor and slow, 
but once you set it up, works very well)


Dovid B escribió:

I liked polycom a lot.

- Original Message -
*From:* Rod Bacon 
*To:* asterisk-users@lists.digium.com

*Sent:* Thursday, February 08, 2007 10:45 AM
*Subject:* [asterisk-users] Best phone for easy provisioning

Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I’ve used CISCO 79XX
phones, and they’re great (but too expensive). I like Grandstream
phones, but their provisioning sucks.

What is everybody else using in large environments where
individual config is not an option?



Rod Bacon

Technical Manager

JASCO Consulting Pty. Ltd.

http://www.jasco.net.au 

Ph. 03 9432 6376

Fax: 03 9432 6378




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[asterisk-users] Transfer -> announce -> ring

[Bill Gibbs]

I am running some Polycom phones and have Auto Answer setup(*51
initiates that when you call an extension)

 

With an attended transfer you can take a call, hit transfer,
*51, announce the call and if the person wants it, complete
the transfer, the call is now on speaker at the end.  This can surprise
people because all of a sudden the call is right there.

 

I know that the attended transfer basically puts the first call on hold,
makes another call, then bridges them together with the 2nd push of the
transfer button, but is there anyway to have it hang up the speaker part
and then _ring_ at the destination?  Basically what would happen if
transferred the call normally without sending the feature to tell the
Polycom to auto answer, but at the same time giving them the ability to
announce it over speaker, all in one fluid motion?

 

I know you could place the call on hold, Intercom over then do a blind
transfer but like all of us here, we have to deal with people who don't
like to push buttons. J

 

I told our people to use Park and then just intercom and the destination
can pick it up at their leisure but I thought I'd raise the question.
That's what this feature is for after all! 

 

So to recap as it works  now:

Call comes in, Transfer, *51, destination Polycom auto answers,
Transfer at source and the call is on speaker

Is it possible to:

Call comes in, Transfer, announce to destination via Auto Answer, then
complete the transfer but have it ring at the destination?

All using the built in Polycom transfer function?

 

 

Bill

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Re: [asterisk-users] Asterisk and 802.11g

[Jason Fuermann]

your asterisk box has to do audio conversion, its getting bogged down

Yuan LIU wrote:
I'm greatly surprised when testing an Asterisk box with 802.11g.  
Here's the topology:


VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
  |
FXO ___ PSTN extension

When I call a VoIP extension on that box (from a VoIP extension), 
voice is good.  But when this box tries to bridge the call with a PSTN 
extension, voice is completely broken.  And it's not because of the 
cheap X100P - when I ping the box, round trip is >4,000 ms, most of 
the time causing timeout.  Once the call hangs up, ping time dropped 
to 1-2 ms.  Ping time started to surge even when FXO is simply ringing.


If VoIP to VoIP extension call uses re-invite (which it did), voice is 
also good in the Console channel.


How can voice traffic stall 802.11g? (I haven't checked, but CODEC is 
likely ulaw.)


Yuan Liu


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Re: [asterisk-users] Best phone for easy provisioning

[Dovid B]

I liked polycom a lot.
  - Original Message - 
  From: Rod Bacon 
  To: asterisk-users@lists.digium.com 
  Sent: Thursday, February 08, 2007 10:45 AM
  Subject: [asterisk-users] Best phone for easy provisioning


  Does anyone have any recommendations for a phone that has easy to 
understand/implement central provisioning? I've used CISCO 79XX phones, and 
they're great (but too expensive). I like Grandstream phones, but their 
provisioning sucks. 

   

  What is everybody else using in large environments where individual config is 
not an option?

   

  

  Rod Bacon

  Technical Manager

  JASCO Consulting Pty. Ltd.

  http://www.jasco.net.au

  Ph. 03 9432 6376

  Fax: 03 9432 6378

  

   



--


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Re: [asterisk-users] Rxfax and Txfax on Asterisk 1.4

[Lee Howard]

Ardjan Zwartjes wrote:


One thing that's important to mention here: We get unpredictable results
if the fax is transmitted entirely over VOIP, if the fax passes a
regular telephony channel once it works fine but if it's purely VOIP,
transmission errors occur. This is probably a timing problem, but we
haven't been able to solve it yet



It's not a timing problem, it's a "that just happens when you do faxing 
over typical VoIP channels" problem.  The reason it happens is because 
of jitter... and more specifically, packet loss.  On UDP communications 
packets are not retransmitted if they get dropped by a router or switch 
or hub.  You can't just take a data stream and chop out small chunks of 
it here and there and expect the receiving end to be okay with it.


Lee.
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Re: [asterisk-users] Re: Asterisk Faxing Support

[tim robinson]

ha ok, I understand now

1) I don't think that Asterisk has any support for meter pulse detection 
on analogue cards.


2) If you already have an ISDN line, why do you not spend the eur 20 on 
a BRI card and do the job properly?  The way you propose you are going 
from ISDN --> Analogue --> digital
If you get a BRI card you do not have anything analogue in the way to 
add echo etc.  You will be MUCH happier with the end result, I promise!


http://www.solwise.co.uk/isdn.htm is what you need - take a look on 
ebay, and you then need to use a bristuffed version of Asterisk.



Rgds
Tim


Patrick wrote:

On Thu, 2007-02-08 at 13:55 +0100, Tomislav Parčina wrote:
  

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...


Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
before t.38 is ever utilised, not even pass-thru.

1.4 Adds support for T.38 pass through only and no other sort of
faxing, the endpoint must support T.38 and you must send your call to
a T.38 gateway and you must not use NAT anywhere in  your network and
you must enable re-invites which could cause CDRs not to reflect the
true details of the call.

Asterisk/Digium also has no interest in any further interest in
expanding T.38 or faxing support in Asterisk.

Steve Underwood and the other fine persons that have helped to develop
the software DSPs and other stuff required for FoIP support also have
no interest in writing any further faxing support for Asterisk (RxFax,
TxFax + the newest span_dsp wont even compile, much less work under
Asterisk any more) probably because they know it will never be
included into the Asterisk code.
  

Someone please tell me this isn't truth.



Afaik it is true that it will not be included in the Asterisk source
because Steve will not disclaim the code to Digium (which he off course
is entitled to). I compiled the latest spandsp (iirc 0.0.3pre27) on a
FC6 box and it compiles fine.

On Steve's website there are versions of app_rxfax and app_txfax for
1.4. Takes some messing around with the 1.4 build system to get them
included but it worked for me last night. Those apps can be found here:

http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/

>From reading this list it seems you are better off using iaxmodem and
Hylafax (I guess that it assuming the fax comes in via TDM on the
Asterisk box). Or check out OpenPBX.org as they have done much work on
T.38 support (visit irc channel #openpbx on freenode.net to talk about
the current status).

Hope this helps.

Regards,
Patrick

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Re: [asterisk-users] Re: Asterisk Faxing Support

[Lee Howard]

Craig Guy wrote:

it wouldn't make business sense for Digium to have code in the free 
distribution that can't be in their commercial distribution.



Yes, I do suspect that Digium sees things this way.

Maybe I'm too much of a free-thinker - too believing in the open-source 
philosophy, but I would like to think that this is not neccesarily 
true.  I would like to think that they could host and support a 
non-disclaimed GPL Asterisk - having features that ABE does not - and 
they would profit from that.  Still, they could ask for disclaimers, and 
undoubtedly many, many people here love Digium enough to do that even if 
they aren't required to do so in order to see their contribution 
integrated upstream.  In the cases where a contributor will not disclaim 
the contribution to Digium then Digium could make some attempt to obtain 
a license for ABE from the contributor (and I expect that this case 
would be extremely rare - but perhaps appropriate for cases like 
spandsp/rxfax/txfax), Digium could write their own rendition of the 
contribution, or ABE could just do without it.


The way I see it, if Asterisk improves or gets some new feature or 
increases its exposure then Digium (and every other business 
commercially involved with Asterisk) will benefit more from that than if 
the improvement had not taken place.


Certainly I think that it's fair to say that some contributions will not 
be disclaimed in the scenario I outlined that would have been disclaimed 
in the present scenario.  I think that depends on how well Digium does 
on keeping the Asterisk users loyal and willing to repay them in kind.  
However, in the end, even if they don't do a good job at that, I think 
that a better Asterisk means a happier Digium... even if that means that 
there is some differences between ABE and Asterisk GPL.


Lee.
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Re: [asterisk-users] Billing pulses

[George Camilleri]

There are two types of ISDN line, Primary Rate Access (PRI) and Basic Rate 
Access (BRI). PRI has 30 (+ 1) channels, BRI has 2 (+1) channels. You are 
talking about BRI which consists of two 64 kbit/s data channels and 1 
signalling channel. In telephony, the two data channels are decoded and used 
as two voice channels. At the end of the decoding process and after passing 
through some interfacing hardware the voice channels end up in an analogue 
device such as a telephone set so that we humans can hear it.


The FXS hardware you invested in can be used for your analogue extensions. 
The FXO hardware is used to interface with analogue telco lines so if you 
want ISDN telco lines you will have to invest in BRI interface cards. 
(Google Asterisk ISDN BRI)


You then ask the telco to include Advice of Charge (AOC) in your ISDN setup. 
The AOC then is included somewhere in the Asterisk CDR, but I don't have 
direct experience of this. You can then get appropriate software to issue 
bills to telephone users.


This is as far as I know and have personal experience of. If anyone can add 
to it it will be appreciated.


George

- Original Message - 
From: "Stefano Corsi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial 
Discussion" 

Sent: Thursday, February 08, 2007 3:13 PM
Subject: Re: [asterisk-users] Billing pulses


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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[asterisk-users] Re: Asterisk Faxing Support

[Justin Newman]

We have considered working on this. T38 is a short term solution, though.

Justin Newman

--

From: Tomislav Par?ina <[EMAIL PROTECTED]>
Subject: [asterisk-users] Re: Asterisk Faxing Support

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
> before t.38 is ever utilised, not even pass-thru.
> 
> 1.4 Adds support for T.38 pass through only and no other sort of
> faxing, the endpoint must support T.38 and you must send your call to
> a T.38 gateway and you must not use NAT anywhere in  your network and
> you must enable re-invites which could cause CDRs not to reflect the
> true details of the call.
> 
> Asterisk/Digium also has no interest in any further interest in
> expanding T.38 or faxing support in Asterisk.
> 
> Steve Underwood and the other fine persons that have helped to develop
> the software DSPs and other stuff required for FoIP support also have
> no interest in writing any further faxing support for Asterisk (RxFax,
> TxFax + the newest span_dsp wont even compile, much less work under
> Asterisk any more) probably because they know it will never be
> included into the Asterisk code.

Someone please tell me this isn't truth.




 

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Get your game face on with the latest PS3 news and previews at Yahoo! Games.
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Re: [asterisk-users] Softphone +Realtime

[Rob Schall]

Is there some way to test this, or to cause the polycom to ignore the
errors, and try back later (unlimited times). The fear I had or
re-registering, was that the softphone would be in use, and the hard
phone would take the number back over. That shouldn't happen until the
number is available (asterisk sets it to 0.0.0.0).

Thoughts?
Rob


Jason Fuermann wrote:
> our Polycoms reregister almost immediately. I think the problem your
> running into is that when the softphone is registered the polycom gets
> some kind of error from asterisk which prevents it from reregistering
>
> Rob Schall wrote:
>> That's what I would have thought. I set the timeout to be 300 secs, but
>> the phone never seems to re-register. We could do a group dial, but like
>> you said, there would be a lot of errors in the log, which we are trying
>> to avoid. Has anyone been able to get a polycom 501 to re-register
>> itself without rebooting it?
>>
>> Rob
>>
>>
>> Chris Bagnall wrote:
>>   
 The first step of getting the phones to log in as their same
 extensions as work is easy and works.
 
   
>>> By definition, I guess that automatically logs out their office phones?
>>>
>>>   
>>> 
 Has anyone tried anything like this? I would like the phones to
 regrab their spot once the softphone is logged out.
 
   
>>> Shouldn't the office phones automatically regrab their spot when they
>>> re-register with the server? If you set the timeout to something fairly
>>> short, it would get around this issue, but introduce another one: the
>>> softphones will be kicked whenever the office phones re-register.
>>>
>>> We have a number of clients doing similar things, but we've taken a slightly
>>> different approach. For example, if we have extensions 201,202 and 203, we
>>> create SIP accounts as follows:
>>> 201
>>> 201-home
>>> 202
>>> 202-home
>>> 203
>>> 203-home
>>>
>>> Then, when connecting calls to those extensions in the dialplan, change
>>> something like:
>>> exten => _2XX,1,Dial(SIP/${EXTEN})
>>>
>>> To:
>>> exten => _2XX,1,Dial(SIP/${EXTEN}&SIP/${EXTEN}-home)
>>>
>>> Hopefully that'll solve the problem. Obviously you'll get lots of errors in
>>> the logs along the lines of "can't find device SIP/202-home" when the
>>> softphones aren't connected, but it shouldn't affect operation.
>>>
>>> Regards,
>>>
>>> Chris
>>>   
>>> 
>>
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[asterisk-users] Transfer

[Thomas Deillon]

Hi all,

 

Me again ... for a new question! Again 

 

Here the scenario:

 

A call B ( A --> B)

B transfer to C ( A --> C)

 

In this case, how can I have the B caller id number and the A caller id
number?

 

Thanks a lot for your help 

 

Thomas

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RE: RE: [asterisk-users] Rxfax and Txfax on Asterisk 1.4

[Ardjan Zwartjes]

That's not nessecerily true, if you install iaxmodem and hylafax on your
asterisk machine you'll use IAX for the internal communication, but
faxes can go out and come in on SIP or whatever you like. 
One thing that's important to mention here: We get unpredictable results
if the fax is transmitted entirely over VOIP, if the fax passes a
regular telephony channel once it works fine but if it's purely VOIP,
transmission errors occur. This is probably a timing problem, but we
haven't been able to solve it yet, still the results are much better
than what we achieved using rx- and txfax.

Kind regards,
Ardjan.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remzi
Semsettin Turer
Sent: donderdag 8 februari 2007 16:18
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: RE: [asterisk-users] Rxfax and Txfax on Asterisk 1.4


This is a solution if your provider is using IAX, but we are stuck with
SIP.

I find it surprising that txfax and rxfax not compiling under 1.4, but
oh well.

Warm Regards,

Remzi Turer

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ardjan
Zwartjes
Sent: Monday, January 29, 2007 2:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Rxfax and Txfax on Asterisk 1.4

> AFAIK the current recommendation is to use HylaFax with something
called iaxmodem.

After having been through a lot of problems with RxFax and TxFax I
completely agree with this statement. Allthough the initial
configuration is a bit complicated, once you have this running you'll
get far better reliability.

Kind regards,
Ardjan Zwartjes. ___
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