Re: [Asterisk-Users] No Audio from Console but mpg123 from shell worksfine.

[Tzafrir Cohen]

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
> k. Creasy
> Sent: Friday, October 14, 2005 10:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] No Audio from Console but mpg123 from shell
> worksfine.
> 
> I get audio from mpg123 at the command line but when I load up asterisk
> and try to get audio from the console it looks like it's working, and
> even pauses like it is playing the file but there is no audio coming
> from the speakers. 

One possibility is that the volume is set to 0. aumix can be handy here.

Try simpler things, like playing wav files with 'play' of sox. 

Do you use ALSA or OSS for sound? What kernel version?

> 
> I have searched and looked through the archives and tried to fix this
> but I have had no success. This is an onboard Intel card (AC'97) and I
> also tried an SB Live card with the same result. 
> 
> -Jonathan
> 
> *
> Asterisk startup: (asterisk -vvvc)
> *
> 
> [chan_oss.so] => (OSS Console Channel Driver)
>   == Parsing '/etc/asterisk/oss.conf': Found
>   == Registered channel type 'Console' (OSS Console Channel Driver)

Asterisk grabs /dev/dsp . I figure you can't play anything at this
point. Though you should get stuck at trying to open it.

> 
> *
> Dial 100:
> *
> 
> *CLI> -- Executing Answer("OSS/dsp", "") in new stack
>  << Console call has been answered >>
> -- Executing Playback("OSS/dsp",
> "tones-that-follow-are-for-the-deaf") in new stack
> -- Playing 'tones-that-follow-are-for-the-deaf' (language 'en')
> 
> *
> *** pause while it plays but no audio ***
> *
> 
> -- Executing Hangup("OSS/dsp", "") in new stack
>   == Spawn extension (default, 100, 3) exited non-zero on 'OSS/dsp'
>  << Hangup on console >>
> 
> *
> Exit asterisk: (ctrl-c which normally I wouldn't do)
> *
> 
> Beginning asterisk shutdown
> Executing last minute cleanups
>   == Destroying musiconhold processes
> Yuck! Error in buffer handling...: Connection reset by peer
> Asterisk cleanly ending (2).
> 
> *
> Run mpg123 (Plays fine): (don't need the /dev/dsp, I was just trying to
> make mpg123 not work to hopefully find out why asterisk doesn't)
> *
> 
> [EMAIL PROTECTED] ~]# mpg123 -d /dev/dsp
> /var/lib/asterisk/mohmp3/TristeAlegriaPromo.mp3
> High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
> Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
> Uses code from various people. See 'README' for more!
> THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
> Title  : 10 - Track 10   Artist: 
> Album  : PROMO   Year  :
> Comment: Genre : Club
> 
> Directory: /var/lib/asterisk/mohmp3/
> Playing MPEG stream from TristeAlegriaPromo.mp3 ...
> Junk at the beginning 49443303
> MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo
> 
> [0:02] Decoding of TristeAlegriaPromo.mp3 finished.
> [EMAIL PROTECTED] ~]#
> 
> *
> Extensions.conf
> *
> 
> exten => 100,1,Answer
> exten => 100,2,Playback(tones-that-follow-are-for-the-deaf)
> exten => 100,3,Hangup
> 
> 
> *
> oss.conf
> *
> ;
> ; Open Sound System Console Driver Configuration File
> ;
> [general]
> ;
> ; Automatically answer incoming calls on the console?  Choose yes if
> ; for example you want to use this as an intercom.
> ;
> autoanswer=yes
> ;
> ; Default context (is overridden with @context syntax)
> ;
> context=default
> ;
> ; Default extension to call
> ;
> extension=s
> ;
> ; Default language
> ;
> ;language=en
> ;
> ; Silence supression can be enabled when sound is over a certain
> threshold.
> ; The value for the threshold should probably be between 500 and 2000 or
> so,
> ; but your mileage may vary.  Use the echo test to evaluate the best
> setting.
> ;silencesuppression = yes
> ;silencethreshold = 1000
> ;
> ; On half-duplex cards, the driver attempts to switch back and forth
> between
> ; read and write modes.  Unfortunately, this fails sometimes on older
> hardware.
> ; To prevent the driver from switching (ie. only play files on your
> speakers),
> ; then set the playbackonly option to yes.  Default is no.  Note this
> option has
> ; no effect on full-duplex cards.
> ;playbackonly=yes
> 
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Re: [Asterisk-Users] sip show peers

[Samy Antoun]

--- Jonathan Lin <[EMAIL PROTECTED]> wrote:
> you get ping time in the status page if your extension.conf has
> qualify=yes

Setup
 #  Device  Location options
200 Sipura  local
210 Sipura  remote   nat=yes qualify=yes
310 eyebeam remote   nat=yes qualify=yes

sip show peers
Name/user Host  Dyn Nat Status
200/200   192.168.1.150  D  Unmonitored
210/210   84.36.36.140   D   N  OK (305 ms)
310/310   71.180.126.60  D   N  Unmonitored

As you can see, ext 310 has qualify=yes and and Unmonitored Status !!!




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Re: [Asterisk-Users] sip show peers

[Jonathan Lin]

you get ping time in the status page if your extension.conf has qualify=yes



Quoting Samy Antoun <[EMAIL PROTECTED]>:


--- Sergey Okhapkin <[EMAIL PROTECTED]> wrote:

Hmm.. What is the output of "sip show users" and "sip show peers"?


sip show users
Username Def.Context  ACL  NAT
200  from-internalNo   No
210  from-internalNo   Always
310  from-internalNo   Always

sip show peers
Name/user Host  Dyn Nat Status
200/200   192.168.1.150  D  Unmonitored
210/210   84.36.36.140   D   N  OK (305 ms)
310/310   71.180.126.60  D   N  Unmonitored





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Re: [Asterisk-Users] What would cause a high memory usage in pbx_spool.c ?

[Eric \"ManxPower\" Wieling]

Walter Klomp wrote:

Hi,

After only 4 days I have 107472352 bytes in 46007 allocations in file 
'pbx_spool.c'


I seem to recall a memory leak in pbx_spool being fixed a few days ago. 
 check the asterisk-cvs mailing list archive on lists.digium.com

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Re: [Asterisk-Users] TDM400 not working -- SOLVED

[Rudolf Ladyzhenskii]

Sorry, I was pretty busy and did not work on my *.

Problem was in zaptel not properly registering driver with udev.
Manually updating udev rules fixed the problem.

Thanks,
Rudolf

- Original Message - 
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>

To: 
Sent: Tuesday, October 11, 2005 6:20 AM
Subject: Re: [Asterisk-Users] TDM400 not working



On Mon, Oct 10, 2005 at 07:25:59PM +1000, Rudolf Ladyzhenskii wrote:

Hi, all

I have installed TDM400 card. I can see it is there (lspci).
But Asterisk does not find it.

phonebox2*CLI> zap show status
No Zaptel interface found.

I assume that driver is not loaded, but I am sure I have installed it (I
compiled zaptel and then re-build asterisk and did make install for both
zaptel and asterisk).

When asterisk is started I get:
Jan  2 06:28:08 WARNING[3473]: chan_zap.c:872 zt_open: Unable to open
'/dev/zap/channel': No such file or directory


The device file does not exast

Jan  2 06:28:08 ERROR[3473]: chan_zap.c:6572 mkintf: Unable to open 
channel

2: No such file or directory
here = 0, tmp->channel = 2, channel = 2
Jan  2 06:28:08 ERROR[3473]: chan_zap.c:9927 setup_zap: Unable to 
register

channel '2'
Jan  2 06:28:08 WARNING[3473]: loader.c:402 __load_resource: chan_zap.so:
load_module failed, returning -1
Jan  2 06:28:08 WARNING[3473]: loader.c:523 load_modules: Loading module
chan_zap.so failed!

Ok, I look in the /dev and I could not find /dev/zap at all! But, there 
is

a /dev/zapchannel character device.


Is that a typo? It should be /dev/zap/channel . Do you use udev? If so,
see README.udev . If not: you need to generate those device files.

Anyway: could you please post the output of:

 lsmod | grep zaptel



Any ideas what can be wrong?

And last question. Does zaptel driver reads configuration file on 
startup?

If so, how do I force the driver to update if config file was changed?


ztcfg loads the configuration to the zaptel module from /etc/zaptel.conf
.

--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] What would cause a high memory usage in pbx_spool.c ?

[Walter Klomp]

Hi,

After only 4 days I have 107472352 bytes in 46007 allocations in file 
'pbx_spool.c'


asterisk*CLI> show memory summary
  180 bytes in 2 allocations in file 'netsock.c'
   12 bytes in 1 allocations in file 'devicestate.c'
 2268 bytes in 1 allocations in file 'jitterbuf.c'
 8160 bytes in 1 allocations in file 'localtime.c'
 1232 bytes in 1 allocations in file 'app_queue.c'
1 bytes in 1 allocations in file 'res_features.c'
   132376 bytes in 2 allocations in file 'res_musiconhold.c'
 3672 bytes in18 allocations in file 'file.c'
 1032 bytes in 2 allocations in file 'enum.c'
  378 bytes in 2 allocations in file 'iax2-parser.c'
  752 bytes in 2 allocations in file 'res_crypto.c'
   21 bytes in 1 allocations in file 'cli.c'
 1348 bytes in 4 allocations in file 'cdr.c'
12984 bytes in19 allocations in file 'chan_iax2.c'
  960 bytes in40 allocations in file 'manager.c'
17647 bytes in18 allocations in file 'app_voicemail.c'
 1456 bytes in 2 allocations in file 'dsp.c'
48456 bytes in 3 allocations in file 'frame.c'
 2203 bytes in15 allocations in file 'channel.c'
42224 bytes in 8 allocations in file 'rtp.c'
   150040 bytes in86 allocations in file 'chan_sip.c'
  230 bytes in 9 allocations in file 'chanvars.c'
25736 bytes in19 allocations in file 'io.c'
 4736 bytes in   177 allocations in file 'asterisk.c'
 4872 bytes in   144 allocations in file 'sched.c'
   513542 bytes in   124 allocations in file 'chan_zap.c'
11907 bytes in   403 allocations in file 'logger.c'
40320 bytes in   140 allocations in file 'loader.c'
 4262 bytes in   265 allocations in file 'res_indications.c'
50839 bytes in   850 allocations in file 'pbx.c'
 7653 bytes in   585 allocations in file 'pbx_config.c'
 2345 bytes in   147 allocations in file 'app_dial.c'
107472352 bytes in 46007 allocations in file 'pbx_spool.c'
  5218074 bytes in 110920 allocations in file 'config.c'
113784270 bytes allocated 160019 units total


And top shows me that asterisk is using 153m virtual memory and 141 res 
memory... Only completely stopping asterisk and starting again will free 
up the memory.


What could be causing this?  I did read something about call files and 
variables in the past, but that was a long time ago. I do have to 
process .call files but not that many...


Could there be a memory leak in pbx_spool.c, how can I help troubleshoot 
that?


Thanks
Walter.
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Re: [Asterisk-Users] ACD calls to busy agents

[Tom Rymes]

I don't know how to make this happen, and I don't even think it is  
really possible given the current Queue app, but this would be a very  
nice feature to have. The queue shouldn't pass a call to an agent if  
they are already on a call from the queue, but an incoming call from  
another internal extension, or even a DID ought to be able to get  
through.


Consider this a feature request?

Tom

On Oct 15, 2005, at 10:04 PM, J Thomas wrote:


One of my friends is facing this problems and I could not find any
solution to that. Hence this post.

In her Asterisk PBX, she has programmed about 10 agents, and  
strategy is
rrmemory. Everything works fine. When an agent has received an ACD  
call,

another call is not presented to him as long as he is on the ACD call.

However when an agent has made an outgoing call, he is still presented
another ACD call when his turn comes. This results in unnecessary  
delay

in answering that call.

Taking out call waiting is not an option, as an agent can also get a
direct dialed call, and he should be able to pick up that call even  
when

he is on another call.

Is there a way so that a busy agent (whether busy because of an  
incoming

call, or outgoing call) is not presented another ACD call?

Thanks,
-- jt

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Re: [Asterisk-Users] Voicemail 2

[FaberK]

Yes, is defined in voicemail.conf too. The context is the glue of the
system, this is what I understood, the way to follow, the arrow that
show the direction to every application.

But the problems, still remain.

2005/10/16, Jason Walker <[EMAIL PROTECTED]>:
> Correct - but is the context defined in voicemail.conf? As mickeymouse? Or
> whatever...?
>
> ;)
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of FaberK
> Sent: Saturday, October 15, 2005 6:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Voicemail 2
>
> well, is just the context.
> You could call it as you prefer, mickeymouse???
> ;o)
> Bye
>
> 2005/10/16, Linc Fessenden <[EMAIL PROTECTED]>:
> > FaberK wrote:
> > > [EMAIL PROTECTED]
> > > ---
> > > Some ideas?
> >
> > Only thing I have that even looks different is [EMAIL PROTECTED]
> >
> > --
> > -Linc Fessenden
> >
> > In the Beginning there was nothing, which exploded - Yeah right...
> >
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>
>
> --
> .:FaberK:.
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RE: [Asterisk-Users] ACD calls to busy agents

[J Thomas]

Setting incominglimit = 1 does not really solve the problem as I had
already mentioned. That practically takes away the call waiting and will
block all incoming calls including direct dialed calls. She does not
want that. Moreover, incominglimit is deprecated too.

-- jt 

On Sat, 2005-10-15 at 22:05, Jason Walker wrote:
> Have you tried the "incominglimit" parameter (or did she)?
> 
> I have found this to work pretty well when limiting the number of calls.
> After monitoring the "full" log, I saw that incoming calls where
> incrementing or decrementing the active call parameter for SIP agents. By
> limiting the number of calls that the phone extension/user can accept at one
> time limited the calls going to an agent.
> 
> I am still trying to figure out how to jump out of the dialplan when a call
> comes into queue -- if anyone has any suggestions for that, it would be
> greatly appreciated.
> 
> But in any event, for similar situations, limiting the number of calls for a
> SIP agent seems to help in the calls coming in on top of another. 
> 
> -Original Message-

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RE: [Asterisk-Users] ACD calls to busy agents

[Jason Walker]

Have you tried the "incominglimit" parameter (or did she)?

I have found this to work pretty well when limiting the number of calls.
After monitoring the "full" log, I saw that incoming calls where
incrementing or decrementing the active call parameter for SIP agents. By
limiting the number of calls that the phone extension/user can accept at one
time limited the calls going to an agent.

I am still trying to figure out how to jump out of the dialplan when a call
comes into queue -- if anyone has any suggestions for that, it would be
greatly appreciated.

But in any event, for similar situations, limiting the number of calls for a
SIP agent seems to help in the calls coming in on top of another. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J Thomas
Sent: Saturday, October 15, 2005 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ACD calls to busy agents

One of my friends is facing this problems and I could not find any solution
to that. Hence this post.

In her Asterisk PBX, she has programmed about 10 agents, and strategy is
rrmemory. Everything works fine. When an agent has received an ACD call,
another call is not presented to him as long as he is on the ACD call.

However when an agent has made an outgoing call, he is still presented
another ACD call when his turn comes. This results in unnecessary delay in
answering that call.

Taking out call waiting is not an option, as an agent can also get a direct
dialed call, and he should be able to pick up that call even when he is on
another call.

Is there a way so that a busy agent (whether busy because of an incoming
call, or outgoing call) is not presented another ACD call?

Thanks,
-- jt

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RE: [Asterisk-Users] Voicemail 2

[Jason Walker]

Correct - but is the context defined in voicemail.conf? As mickeymouse? Or
whatever...?

;)

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FaberK
Sent: Saturday, October 15, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail 2

well, is just the context.
You could call it as you prefer, mickeymouse???
;o)
Bye

2005/10/16, Linc Fessenden <[EMAIL PROTECTED]>:
> FaberK wrote:
> > [EMAIL PROTECTED]
> > ---
> > Some ideas?
>
> Only thing I have that even looks different is [EMAIL PROTECTED]
>
> --
> -Linc Fessenden
>
> In the Beginning there was nothing, which exploded - Yeah right...
>
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--
.:FaberK:.
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[Asterisk-Users] Looking for Info on OH323

[Obelix]

I have compiled the OH323 module for my system.

When can I find some info on how to properly configure it?

I haven't read any info for its configuration, and I need some starting info.

Were do I start?

Obelix



This message was sent using IMP, the Internet Messaging Program.

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[Asterisk-Users] ACD calls to busy agents

[J Thomas]

One of my friends is facing this problems and I could not find any
solution to that. Hence this post.

In her Asterisk PBX, she has programmed about 10 agents, and strategy is
rrmemory. Everything works fine. When an agent has received an ACD call,
another call is not presented to him as long as he is on the ACD call.

However when an agent has made an outgoing call, he is still presented
another ACD call when his turn comes. This results in unnecessary delay
in answering that call.

Taking out call waiting is not an option, as an agent can also get a
direct dialed call, and he should be able to pick up that call even when
he is on another call.

Is there a way so that a busy agent (whether busy because of an incoming
call, or outgoing call) is not presented another ACD call?

Thanks,
-- jt

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Re: [Asterisk-Users] Voicemail 2

[FaberK]

well, is just the context.
You could call it as you prefer, mickeymouse???
;o)
Bye

2005/10/16, Linc Fessenden <[EMAIL PROTECTED]>:
> FaberK wrote:
> > [EMAIL PROTECTED]
> > ---
> > Some ideas?
>
> Only thing I have that even looks different is
> [EMAIL PROTECTED]
>
> --
> -Linc Fessenden
>
> In the Beginning there was nothing, which exploded - Yeah right...
>
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.:FaberK:.
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Re: [Asterisk-Users] Voicemail 2

[FaberK]

First answer:
the envelope came back to me!
Yesterday I've added the line:
notifymimetype=text/plain
into sip.conf, so no more envelope just text into logs.
Is just an answer to you, Linc.

Nothing for me, yet.

2005/10/16, FaberK <[EMAIL PROTECTED]>:
> I've got it too until yesterday!!!
> Now no more envelope either.
> This is from extensions.conf:
> ---
> exten => 221,1,Dial(SIP/221,20,tr)
> exten => 221,2,Voicemail(u${EXTEN})
> exten => 221,102,Voicemail(b${EXTEN})
> exten => 221,103,Hangup
> ---
> this is from sip.conf:
> ---
> [221]
> type=friend
> username=221
> secret=221
> callerid="221" <221>
> fromuser=221
> accountcode=221
> context=local
> host=dynamic
> dtmfmode=rfc2833
> nat=yes
> qualify=yes
> Port=5060
> [EMAIL PROTECTED]
> Disallow=all
> Allow=gsm
> Allow=ulaw
> Allow=alaw
> ---
> Some ideas?
>
> 2005/10/16, Linc Fessenden <[EMAIL PROTECTED]>:
> > FaberK wrote:
> > > Hi list,
> > > I'm trying, as usual, to set up voicemail.
> > > It works, but signaling to phones, doesn't.
> > > Into XLite logs, I have:
> > > --
> > > Messages-Waiting: yes
> > >
> > > Message-Account: sip:[EMAIL PROTECTED]
> > >
> > > Voice-Message: 1/0 (0/0)
> > > --
> > > but nothing appear on the XLite screen.
> > > So, I understand that I'm able to send the right signal, but something
> > > is still wrong.
> > > Ideas?
> > >
> > > Thanks in advance
> > > --
> > > .:FaberK:.
> >
> > I have something similar.  I have the little mail envelope on the screen
> > of xten-xlite, but can't figure out how to clear it off.
> >
> > --
> > -Linc Fessenden
> >
> > In the Beginning there was nothing, which exploded - Yeah right...
> >
> > ___
> > --Bandwidth and Colocation sponsored by Easynews.com --
> >
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> .:FaberK:.
>


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.:FaberK:.
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Re: [Asterisk-Users] Voicemail 2

[Linc Fessenden]

FaberK wrote:

[EMAIL PROTECTED]
---
Some ideas?


Only thing I have that even looks different is
[EMAIL PROTECTED]

--
-Linc Fessenden

In the Beginning there was nothing, which exploded - Yeah right...

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RE: [Asterisk-Users] No Audio from Console but mpg123 from shell worksfine.

[Jonathan k. Creasy]

Anyone have anything on this? (I'm sure someone will complain about me
bringing it up again, chill out...)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, October 14, 2005 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] No Audio from Console but mpg123 from shell
worksfine.

I get audio from mpg123 at the command line but when I load up asterisk
and try to get audio from the console it looks like it's working, and
even pauses like it is playing the file but there is no audio coming
from the speakers. 

I have searched and looked through the archives and tried to fix this
but I have had no success. This is an onboard Intel card (AC'97) and I
also tried an SB Live card with the same result. 

-Jonathan

*
Asterisk startup: (asterisk -vvvc)
*

[chan_oss.so] => (OSS Console Channel Driver)
  == Parsing '/etc/asterisk/oss.conf': Found
  == Registered channel type 'Console' (OSS Console Channel Driver)

*
Dial 100:
*

*CLI> -- Executing Answer("OSS/dsp", "") in new stack
 << Console call has been answered >>
-- Executing Playback("OSS/dsp",
"tones-that-follow-are-for-the-deaf") in new stack
-- Playing 'tones-that-follow-are-for-the-deaf' (language 'en')

*
*** pause while it plays but no audio ***
*

-- Executing Hangup("OSS/dsp", "") in new stack
  == Spawn extension (default, 100, 3) exited non-zero on 'OSS/dsp'
 << Hangup on console >>

*
Exit asterisk: (ctrl-c which normally I wouldn't do)
*

Beginning asterisk shutdown
Executing last minute cleanups
  == Destroying musiconhold processes
Yuck! Error in buffer handling...: Connection reset by peer
Asterisk cleanly ending (2).

*
Run mpg123 (Plays fine): (don't need the /dev/dsp, I was just trying to
make mpg123 not work to hopefully find out why asterisk doesn't)
*

[EMAIL PROTECTED] ~]# mpg123 -d /dev/dsp
/var/lib/asterisk/mohmp3/TristeAlegriaPromo.mp3
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.
Uses code from various people. See 'README' for more!
THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
Title  : 10 - Track 10   Artist: 
Album  : PROMO   Year  :
Comment: Genre : Club

Directory: /var/lib/asterisk/mohmp3/
Playing MPEG stream from TristeAlegriaPromo.mp3 ...
Junk at the beginning 49443303
MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo

[0:02] Decoding of TristeAlegriaPromo.mp3 finished.
[EMAIL PROTECTED] ~]#

*
Extensions.conf
*

exten => 100,1,Answer
exten => 100,2,Playback(tones-that-follow-are-for-the-deaf)
exten => 100,3,Hangup


*
oss.conf
*
;
; Open Sound System Console Driver Configuration File
;
[general]
;
; Automatically answer incoming calls on the console?  Choose yes if
; for example you want to use this as an intercom.
;
autoanswer=yes
;
; Default context (is overridden with @context syntax)
;
context=default
;
; Default extension to call
;
extension=s
;
; Default language
;
;language=en
;
; Silence supression can be enabled when sound is over a certain
threshold.
; The value for the threshold should probably be between 500 and 2000 or
so,
; but your mileage may vary.  Use the echo test to evaluate the best
setting.
;silencesuppression = yes
;silencethreshold = 1000
;
; On half-duplex cards, the driver attempts to switch back and forth
between
; read and write modes.  Unfortunately, this fails sometimes on older
hardware.
; To prevent the driver from switching (ie. only play files on your
speakers),
; then set the playbackonly option to yes.  Default is no.  Note this
option has
; no effect on full-duplex cards.
;playbackonly=yes

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Re: [Asterisk-Users] Callerid on t1 lines

[C F]

What is the adit 600 doing? FXO? FXS? how you connected to the PSTN?
I got an Adit 600 with both FXO and FXS as well as a PRI and I'm
getting CallerID on all three.

On 10/14/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hello All,
> Just a question, I have an adit600 and I am looking for a way to pull
> the incoming cid into asterisk.
>
> Does anyone know if this is just not possible via t1? Or is it only
> available on PRI?
>
> Thanks,
> Greg
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Re: [Asterisk-Users] Voicemail 2

[FaberK]

I've got it too until yesterday!!!
Now no more envelope either.
This is from extensions.conf:
---
exten => 221,1,Dial(SIP/221,20,tr)
exten => 221,2,Voicemail(u${EXTEN})
exten => 221,102,Voicemail(b${EXTEN})
exten => 221,103,Hangup
---
this is from sip.conf:
---
[221]
type=friend
username=221
secret=221
callerid="221" <221>
fromuser=221
accountcode=221
context=local
host=dynamic
dtmfmode=rfc2833
nat=yes
qualify=yes
Port=5060
[EMAIL PROTECTED]
Disallow=all
Allow=gsm
Allow=ulaw
Allow=alaw
---
Some ideas?

2005/10/16, Linc Fessenden <[EMAIL PROTECTED]>:
> FaberK wrote:
> > Hi list,
> > I'm trying, as usual, to set up voicemail.
> > It works, but signaling to phones, doesn't.
> > Into XLite logs, I have:
> > --
> > Messages-Waiting: yes
> >
> > Message-Account: sip:[EMAIL PROTECTED]
> >
> > Voice-Message: 1/0 (0/0)
> > --
> > but nothing appear on the XLite screen.
> > So, I understand that I'm able to send the right signal, but something
> > is still wrong.
> > Ideas?
> >
> > Thanks in advance
> > --
> > .:FaberK:.
>
> I have something similar.  I have the little mail envelope on the screen
> of xten-xlite, but can't figure out how to clear it off.
>
> --
> -Linc Fessenden
>
> In the Beginning there was nothing, which exploded - Yeah right...
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


--
.:FaberK:.
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Re: [Asterisk-Users] Voicemail 2

[Linc Fessenden]

FaberK wrote:

Hi list,
I'm trying, as usual, to set up voicemail.
It works, but signaling to phones, doesn't.
Into XLite logs, I have:
--
Messages-Waiting: yes

Message-Account: sip:[EMAIL PROTECTED]

Voice-Message: 1/0 (0/0)
--
but nothing appear on the XLite screen.
So, I understand that I'm able to send the right signal, but something
is still wrong.
Ideas?

Thanks in advance
--
.:FaberK:.


I have something similar.  I have the little mail envelope on the screen 
of xten-xlite, but can't figure out how to clear it off.


--
-Linc Fessenden

In the Beginning there was nothing, which exploded - Yeah right...

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[Asterisk-Users] Voicemail 2

[FaberK]

Hi list,
I'm trying, as usual, to set up voicemail.
It works, but signaling to phones, doesn't.
Into XLite logs, I have:
--
Messages-Waiting: yes

Message-Account: sip:[EMAIL PROTECTED]

Voice-Message: 1/0 (0/0)
--
but nothing appear on the XLite screen.
So, I understand that I'm able to send the right signal, but something
is still wrong.
Ideas?

Thanks in advance
--
.:FaberK:.
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RE: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

[Rich Adamson]

The appendices are in the book. :) 
Apparently only missing in the pdf.


> Is appendix A and B missing?
> 
> > -Original Message-
> >  
> > Leif Madsen a écrit :
> > 
> > >Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk 
> > >Documentation Project, in conjunction with O'Reilly Media 
> > are pleased 
> > >to announce the official release of Asterisk: The Future of 
> > Telephony 
> > >on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.
> > >
> > >In the true spirit of Open Source, the authors and O'Reilly 
> > Media have 
> > >published the book under the open, Creative Commons license, 
> > allowing 
> > >the book in its entirity to be freely distributed.


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Re: [Asterisk-Users] res_perl - Compiling error

[Tom Vile]

Maybe you don't have libc6-dev installed?
On 10/15/05, Brent August Torrenga <[EMAIL PROTECTED]> wrote:
Having trouble running make on res_perl:[EMAIL PROTECTED] res_perl]# makeperl -MExtUtils::Embed -e xsinitgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DMULTIPLICITY  -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS
-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE-D_FILE_OFFSET_BITS=64-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE   -c -o perlxsi.operlxsi.cgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-
1.0.9/-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"
-DMULTIPLICITY  -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE-D_FILE_OFFSET_BITS=64-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE   -c -o res_perl.o
res_perl.cgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk
\"-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=
\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DMULTIPLICITY  -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE   -c -o apihelp.oapihelp.cgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"-DMULTIPLICITY  -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS
-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE-D_FILE_OFFSET_BITS=64-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE   -c -oAstAPIBase.o AstAPIBase.cgcc -Wall -fPIC -shared -Xlinker -x -o res_perl.so 
perlxsi.o res_perl.oapihelp.o -Wl,-E-L/usr/local/lib /usr/local/lib/perl5/5.8.7/-linux-thread-multi/auto/DynaLoader/DynaLoader.a -L/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE
-lperl -lnsl -ldl -lm -lcrypt -lutil -lpthread -lc -lnsl -lndbm -lgdbm
-ldl -lm -lcrypt -lutil -lc  AstAPIBase.o/usr/bin/ld: cannot find -lndbmcollect2: ld returned 1 exit statusmake: *** [res_perl.so] Error 1[EMAIL PROTECTED] res_perl]#___
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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[Asterisk-Users] res_perl - Compiling error

[Brent August Torrenga]

Having trouble running make on res_perl:

[EMAIL PROTECTED] res_perl]# make
perl -MExtUtils::Embed -e xsinit
gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/
-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk
\"-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=
\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"
-DMULTIPLICITY  -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS
-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64
-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE   -c -o perlxsi.o
perlxsi.c
gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/
-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk
\"-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=
\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"
-DMULTIPLICITY  -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS
-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64
-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE   -c -o res_perl.o
res_perl.c
gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/
-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk
\"-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=
\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"
-DMULTIPLICITY  -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS
-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64
-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE   -c -o apihelp.o
apihelp.c
gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/
-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\"
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk
\"-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=
\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\"
-DMULTIPLICITY  -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS
-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE
-D_FILE_OFFSET_BITS=64
-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE   -c -o
AstAPIBase.o AstAPIBase.c
gcc -Wall -fPIC -shared -Xlinker -x -o res_perl.so perlxsi.o res_perl.o
apihelp.o -Wl,-E
-L/usr/local/lib 
/usr/local/lib/perl5/5.8.7/-linux-thread-multi/auto/DynaLoader/DynaLoader.a 
-L/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -lperl -lnsl -ldl -lm 
-lcrypt -lutil -lpthread -lc -lnsl -lndbm -lgdbm -ldl -lm -lcrypt -lutil -lc  
AstAPIBase.o
/usr/bin/ld: cannot find -lndbm
collect2: ld returned 1 exit status
make: *** [res_perl.so] Error 1
[EMAIL PROTECTED] res_perl]#


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RE: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

[Terry Vaught]

Is appendix A and B missing?

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jean-Michel Hiver
> Sent: Saturday, October 15, 2005 5:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Discussions regarding The Asterisk Documentation Project
> Subject: Re: [Asterisk-Users] You ASKED for an Asterisk 
> book,you GOT an Asterisk book!
> 
>  
> Leif Madsen a écrit :
> 
> >Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk 
> >Documentation Project, in conjunction with O'Reilly Media 
> are pleased 
> >to announce the official release of Asterisk: The Future of 
> Telephony 
> >on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.
> >
> >In the true spirit of Open Source, the authors and O'Reilly 
> Media have 
> >published the book under the open, Creative Commons license, 
> allowing 
> >the book in its entirity to be freely distributed.
> >  
> >
> Kudos for this. Brilliant!
> 
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Re: [Asterisk-Users] sip show peers

[Samy Antoun]

--- Sergey Okhapkin <[EMAIL PROTECTED]> wrote:
> Hmm.. What is the output of "sip show users" and "sip show peers"?

sip show users
Username Def.Context  ACL  NAT
200  from-internalNo   No
210  from-internalNo   Always
310  from-internalNo   Always

sip show peers
Name/user Host  Dyn Nat Status
200/200   192.168.1.150  D  Unmonitored
210/210   84.36.36.140   D   N  OK (305 ms)
310/310   71.180.126.60  D   N  Unmonitored





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Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

[Jean-Michel Hiver]

Leif Madsen a écrit :


Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.

In the true spirit of Open Source, the authors and O'Reilly Media have
published the book under the open, Creative Commons license, allowing
the book in its entirity to be freely distributed.
 


Kudos for this. Brilliant!

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Re: [Asterisk-Users] Busy not jumping n + 101 anymore

[Andrew Kohlsmith]

On Saturday 15 October 2005 16:18, [EMAIL PROTECTED] wrote:
> Well - number 1 - it IS - CVS HEAD.

Agreed.

> Next - I always run "make samples".  In the /etc/asterisk/ directory it
> renames all your old config files that have changed to *.old.
>
> So long as you don't stop and restart Asterisk - its fine.  Just diff the
> two files and you get to see the differences.

A diff over my config files and the samples would be useless.  Better to just 
read over the samples to see if there's anything obvious.

HOWEVER.  There's a changelog that should have this data in it, and the CVS 
logs should also have it, but more importantly, the upgrade.txt should have 
had this.  It's just an oversight, one that I hope Digium'll fix shortly.

> It's called system administration.  One day I'll set it up with a
> script, but right now it isn't bad or too much work.  But I do update
> almost everyday.

Nice passive-agressive attack there.  System administration has nothing to do 
with what making samples and diff'ing.  Keeping a log of changes you've 
made... now that is indicative of good administration.

-A.
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Re: [Asterisk-Users] Quad BRI with Fedora, anyone?

[Harald Holzer]

You can find RPMS with bristuff included at:
http://www.laimbock.com/asterisk/

the are compiled for centos.
rebuilding the SRPMS under FC3 work without a problem.

> We have a QuadBRI ISDN card from Digium. We would like to make it work
> with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of
> bristuff from the Digium homepage fails, both the stable version with
> asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD.
>
> Has anybody here succeeded to make this work? Or could we even be so
> lucky that somebody made RPMs for this?
>
> Lars Dybdahl.
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Re: [Asterisk-Users] sip show peers

[Sergey Okhapkin]

Hmm.. What is the output of "sip show users" and "sip show peers"?

On Sat, 2005-10-15 at 12:30 -0700, Samy Antoun wrote:


--- Sergey Okhapkin <[EMAIL PROTECTED]> wrote:
> Are the devices at 200 and 310 set up to register with your asterisk?

Yes, they are registered and I can call them
 




		
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RE: [Asterisk-Users] You ASKED for an Asterisk book, you GOT anAsterisk book!

[Cheyenne]

I´m a new user.I´m reading it seams to be very good.
Great job. It´s realy the book i as tryng to find!

Regards!


Cumprimentos,
André Rodrigues 
 
Grupo Paulo Serra & Irmãos, Lda. 
Direcção de Sistemas de Informação 
Tel.: +351 25237 (ext: 296)
Fax: +351 252313483 
Telem : +351 964245524
E-mail :[EMAIL PROTECTED]
Website: www.cheyenne-pt.com

-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Mr. James W.
Laferriere
Enviada: sábado, 15 de Outubro de 2005 20:46
Para: Leif Madsen; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Discussions regarding The Asterisk Documentation Project
Assunto: Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT
anAsterisk book!

Hello Leif ,  The appendices A & B are missing from the zip file
available at the location mentioned below .  Is there some reason of
copyright that is not mentioned here ?  Tia ,  JimL

On Sat, 15 Oct 2005, Leif Madsen wrote:
> Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk 
> Documentation Project, in conjunction with O'Reilly Media are pleased 
> to announce the official release of Asterisk: The Future of Telephony 
> on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.
>
> In the true spirit of Open Source, the authors and O'Reilly Media have 
> published the book under the open, Creative Commons license, allowing 
> the book in its entirity to be freely distributed.
>
> Asterisk: The Future of Telephony is now freely available, for 
> download in PDF form, from the Asterisk Documentation Project website 
> located at http://www.asteriskdocs.org. On the left hand side, click 
> on "Read the book online!" for a copy.
>
> The authors would like to thank O'Reilly Media for having the vision 
> to understand how significant it is for the Asterisk community to have 
> a book freely available, thereby lowering the barrier of entry for 
> those new to Asterisk, and to give back to a project that has given us 
> all so much.
>
> I would personally like to thank Jared Smith, Jim van Meggelen, 
> Michael Loukides (our editor) and the entire O'Reilly Media staff.
>
> The book is currently shipping, and should be available at all major 
> book stores in paperback, and also online from 
> http://www.oreilly.com/catalog/asterisk/ and other online outlets.
>
> Thanks, and we hope you enjoy reading it as much as we enjoyed writing it!
>
> PS: If the Asterisk Documentation Project website becomes slow due to 
> the number of people accessing it at once, we appoligize and 
> appreciate your patience. For those of you who are able to obtain the 
> full copy, please consider helping us out by creating mirrors and 
> torrents and posting them to the list by replying to this thread.
> Thanks!
>
> --
> Leif Madsen - http://www.leifmadsen.com 
> http://www.oreilly.com/catalog/asterisk
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-- 
+--+
| James   W.   Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. |  Give me Linux  |
| [EMAIL PROTECTED] | Billings , MT. 59105 |   only  on  AXP |
+--+
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Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

[Mr. James W. Laferriere]

Hello Leif ,  The appendices A & B are missing from the zip file
available at the location mentioned below .  Is there some reason of
copyright that is not mentioned here ?  Tia ,  JimL

On Sat, 15 Oct 2005, Leif Madsen wrote:

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.

In the true spirit of Open Source, the authors and O'Reilly Media have
published the book under the open, Creative Commons license, allowing
the book in its entirity to be freely distributed.

Asterisk: The Future of Telephony is now freely available, for
download in PDF form, from the Asterisk Documentation Project website
located at http://www.asteriskdocs.org. On the left hand side, click
on "Read the book online!" for a copy.

The authors would like to thank O'Reilly Media for having the vision
to understand how significant it is for the Asterisk community to have
a book freely available, thereby lowering the barrier of entry for
those new to Asterisk, and to give back to a project that has given us
all so much.

I would personally like to thank Jared Smith, Jim van Meggelen,
Michael Loukides (our editor) and the entire O'Reilly Media staff.

The book is currently shipping, and should be available at all major
book stores in paperback, and also online from
http://www.oreilly.com/catalog/asterisk/ and other online outlets.

Thanks, and we hope you enjoy reading it as much as we enjoyed writing it!

PS: If the Asterisk Documentation Project website becomes slow due to
the number of people accessing it at once, we appoligize and
appreciate your patience. For those of you who are able to obtain the
full copy, please consider helping us out by creating mirrors and
torrents and posting them to the list by replying to this thread.
Thanks!

--
Leif Madsen - http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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--
+--+
| James   W.   Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. |  Give me Linux  |
| [EMAIL PROTECTED] | Billings , MT. 59105 |   only  on  AXP |
+--+
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Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

[tim panton]

On 15 Oct 2005, at 19:58, Leif Madsen wrote:


Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.

In the true spirit of Open Source, the authors and O'Reilly Media have
published the book under the open, Creative Commons license, allowing
the book in its entirity to be freely distributed.

Asterisk: The Future of Telephony is now freely available, for
download in PDF form, from the Asterisk Documentation Project website
located at http://www.asteriskdocs.org. On the left hand side, click
on "Read the book online!" for a copy.

The authors would like to thank O'Reilly Media for having the vision
to understand how significant it is for the Asterisk community to have
a book freely available, thereby lowering the barrier of entry for
those new to Asterisk, and to give back to a project that has given us
all so much.

I would personally like to thank Jared Smith, Jim van Meggelen,
Michael Loukides (our editor) and the entire O'Reilly Media staff.

The book is currently shipping, and should be available at all major
book stores in paperback, and also online from
http://www.oreilly.com/catalog/asterisk/ and other online outlets.

Thanks, and we hope you enjoy reading it as much as we enjoyed  
writing it!


PS: If the Asterisk Documentation Project website becomes slow due to
the number of people accessing it at once, we appoligize and
appreciate your patience. For those of you who are able to obtain the
full copy, please consider helping us out by creating mirrors and
torrents and posting them to the list by replying to this thread.
Thanks!



No, thank you! I've read the book and it is very good.
Just what asterisk needed.

I've mirrored it on our website at
http://www.westhawk.co.uk/resources/AsteriskTFOT.zip


Tim



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Re: [Asterisk-Users] Quad BRI with Fedora, anyone?

[stoffell]

On 10/15/05, Lars Dybdahl <[EMAIL PROTECTED]> wrote:
It seems that the bristuff source from junghanns.com wasn't written
for gcc 4, which is the one included in FC4, and I have seen somedescriptions on making zaphfc compile, but there are more problemsthan just that one. Also, RPMs would reduce the amount of time spenton making this work significantly.

What kernel are you using?
What version of bristuff are you using? (latest is bristuff-0.2.0-RC8p.tar.gz)
Where do you encounter errors, and wich errors do you get?


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Re: [Asterisk-Users] Quad BRI with Fedora, anyone?

[Dave Cotton]

On Sat, 2005-10-15 at 21:35 +0200, Lars Dybdahl wrote:
> I had a typo in my original mail - of course it's from junghanns.net,
> not from digium, and so is the bristuff, that I downloaded.
> 
> I'm not searching for solutions on how to make it compile - I'm just
> trying to find out if anybody succeeded in having it work on FC3 and
> FC4, and if yes, if there are RPMs available.
> 
> It seems that the bristuff source from junghanns.com wasn't written
> for gcc 4, which is the one included in FC4, and I have seen some
> descriptions on making zaphfc compile, but there are more problems
> than just that one. Also, RPMs would reduce the amount of time spent
> on making this work significantly.

Well all I can say is that I've compiled and loaded it without problem
on Mandriva Cooker with gcc4 and also on a 64 bit system.

-- 
Dave Cotton <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Busy not jumping n + 101 anymore

[brett]

On 10/15/2005, "John Novack" <[EMAIL PROTECTED]> wrote:
>Eric "ManxPower" Wieling wrote:
>
>> Andrew Kohlsmith wrote:
>>
>>> On Friday 14 October 2005 20:50, Eric "ManxPower" Wieling wrote:
>>>
 Oddly enough, I believe it's mentioned in UPGRADE.txt.
>>>
>>> Care to tell us where?  I just checked my CVS HEAD copy of UPGRADE.txt.
>>
>>
>> Sorry, it's in asterisk/configs/extensions.conf.sample
>
> Which isn't even produced if one doesn't  "make samples"
>
> What backwards thinking put the information there, and in addition
> changed the way jumps used to work as the default?
> If more time were spent on fixing things that were broken, and making
> the interface to the existing PSTN analog lines work smoother there
> might be more acceptance.
>
> JMO

Well - number 1 - it IS - CVS HEAD.

Next - I always run "make samples".  In the /etc/asterisk/ directory it
renames all your old config files that have changed to *.old.

So long as you don't stop and restart Asterisk - its fine.  Just diff the
two files and you get to see the differences.

Now that seems a 'funny' thing to do as you are thinking - heck ALL my
configs are 'different' - as are mine.  But I keep a subdirectory full
of MY config files (just in case) and I also keep a subdirectory full of
the LAST updates config files just to compare against.  After seeing
what's new - fixing MY config files - moving the NEW ones to the latest
subdirectory - moving MY config files back to the working directory -
THEN I restart Asterisk.

It's called system administration.  One day I'll set it up with a
script, but right now it isn't bad or too much work.  But I do update
almost everyday.

Brett
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Re: [Asterisk-Users] Attempted to delete nonexistent schedule entry...

[brett]

On 10/15/2005, "J. Iddings" <[EMAIL PROTECTED]> wrote:
>I'm also having this issue. Everything seems to work, but it's an
>unnerving error. Any thoughts?
>
>Jimmy wrote:
>> I just upgraded my test Asterisk box to the latest CVS HEAD.  "show
>> version" only shows  "Asterisk CVS HEAD built by rootetc", with no
>> date or version number.  I downloaded  this version on Monday, Oct 3.
>> About once every minute, I get this while at the CLI> prompt:
>>
>> sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule
>> entry 1!
>>
>> This only appeared after updating.  All functions seem normal, other
>> than these messages. Phones work, auto-attendant works, voicemail works,
>> etc.  What's going on?

OK - I been wrong so many times this week - it ain't funny...

But - I think - this part of the scheduling change to the registration
stuff.

In one update, when a remote phone/system stopped responding to qualify
attempts, the system would stop trying to verify the connection. 
Forever.
Not exactly a 'good thing'.  It would tell you that by saying Forever
but
still not good.

Then an update added some stuff to ?iax.conf? like:
;qualify=yes
;qualifysmoothing = yes
;qualifyfreqok = 12
;qualifyfreqnotok = 3

to modify how and when the system would retry these connections.

During the time between the first and second update, I would get these
messages when I did an iax2 reload.  It had stopped trying to qualify the
connection - and then the reload would start it backup.  It would
'inform'
me with the 'attempted to delete nonexistant schedule entry' because the
time of the next scheduled event was no longer active.

So in essence - it is a warning and not an error.

Brett
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Re: [Asterisk-Users] Quad BRI with Fedora, anyone?

[Lars Dybdahl]

I had a typo in my original mail - of course it's from junghanns.net,
not from digium, and so is the bristuff, that I downloaded.

I'm not searching for solutions on how to make it compile - I'm just
trying to find out if anybody succeeded in having it work on FC3 and
FC4, and if yes, if there are RPMs available.

It seems that the bristuff source from junghanns.com wasn't written
for gcc 4, which is the one included in FC4, and I have seen some
descriptions on making zaphfc compile, but there are more problems
than just that one. Also, RPMs would reduce the amount of time spent
on making this work significantly.

I hope you can forgive my typo in my original mail and give me a hint or two :-)

Lars Dybdahl


On 10/15/05, Dave Cotton <[EMAIL PROTECTED]> wrote:
> Perhaps if you stated more clearly the actual errors some one might be
> able to help you.  What worries me, is you don't seem to know where you
> got the source from.
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Re: [Asterisk-Users] sip show peers

[Samy Antoun]

--- Sergey Okhapkin <[EMAIL PROTECTED]> wrote:
> Are the devices at 200 and 310 set up to register with your asterisk?

Yes, they are registered and I can call them
 





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Re: [Asterisk-Users] sip show peers

[Sergey Okhapkin]

Are the devices at 200 and 310 set up to register with your asterisk?

On Sat, 2005-10-15 at 11:42 -0700, Samy Antoun wrote:


Hi,

I have 3 SIP extensions, setup as follows:
 #  Device  Location options
200 Sipura  local
210 Sipura  remote   nat=yes qualify=yes
310 eyebeam remote   nat=yes qualify=yes

This is the result of sip show peers:
Name/user Host  Dyn Nat Status
200/200   192.168.1.150  D  Unmonitored
210/210   84.36.36.140   D   N  OK (305 ms)
310/310   71.180.126.60  D   N  Unmonitored

Does anyone know why ext 210 the only one has a ping
status "OK (305 ms)" and the others are "Unmonitored"

Regards



		
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Re: [Asterisk-Users] Busy not jumping n + 101 anymore

[John Novack]

Eric "ManxPower" Wieling wrote:


Andrew Kohlsmith wrote:


On Friday 14 October 2005 20:50, Eric "ManxPower" Wieling wrote:


Oddly enough, I believe it's mentioned in UPGRADE.txt.




Care to tell us where?  I just checked my CVS HEAD copy of UPGRADE.txt.



Sorry, it's in asterisk/configs/extensions.conf.sample


Which isn't even produced if one doesn't  "make samples"

What backwards thinking put the information there, and in addition 
changed the way jumps used to work as the default?
If more time were spent on fixing things that were broken, and making 
the interface to the existing PSTN analog lines work smoother there 
might be more acceptance.


JMO

JN

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Re: [Asterisk-Users] Attempted to delete nonexistent schedule entry...

[J. Iddings]

I'm also having this issue. Everything seems to work, but it's an
unnerving error. Any thoughts?

Jimmy wrote:
> I just upgraded my test Asterisk box to the latest CVS HEAD.  "show
> version" only shows  "Asterisk CVS HEAD built by rootetc", with no
> date or version number.  I downloaded  this version on Monday, Oct 3.
> About once every minute, I get this while at the CLI> prompt:
> 
> sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule
> entry 1!
> 
> This only appeared after updating.  All functions seem normal, other
> than these messages. Phones work, auto-attendant works, voicemail works,
> etc.  What's going on?

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[Asterisk-Users] DID on analog line

[Apu Islam]

Can someone tell me how I can test DID on analog lines ? I have the WCfxo clone and would like to try DID on it.
Can my telco (qwest) provide that service ?
 
 
-apu
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[Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!

[Leif Madsen]

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.

In the true spirit of Open Source, the authors and O'Reilly Media have
published the book under the open, Creative Commons license, allowing
the book in its entirity to be freely distributed.

Asterisk: The Future of Telephony is now freely available, for
download in PDF form, from the Asterisk Documentation Project website
located at http://www.asteriskdocs.org. On the left hand side, click
on "Read the book online!" for a copy.

The authors would like to thank O'Reilly Media for having the vision
to understand how significant it is for the Asterisk community to have
a book freely available, thereby lowering the barrier of entry for
those new to Asterisk, and to give back to a project that has given us
all so much.

I would personally like to thank Jared Smith, Jim van Meggelen,
Michael Loukides (our editor) and the entire O'Reilly Media staff.

The book is currently shipping, and should be available at all major
book stores in paperback, and also online from
http://www.oreilly.com/catalog/asterisk/ and other online outlets.

Thanks, and we hope you enjoy reading it as much as we enjoyed writing it!

PS: If the Asterisk Documentation Project website becomes slow due to
the number of people accessing it at once, we appoligize and
appreciate your patience. For those of you who are able to obtain the
full copy, please consider helping us out by creating mirrors and
torrents and posting them to the list by replying to this thread.
Thanks!

--
Leif Madsen - http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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Re: [Asterisk-Users] Quad BRI with Fedora, anyone?

[Dave Cotton]

On Sat, 2005-10-15 at 14:29 +0200, Lars Dybdahl wrote:
> We have a QuadBRI ISDN card from Digium. We would like to make it work
> with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of
> bristuff from the Digium homepage fails, both the stable version with
> asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD.
> 
> Has anybody here succeeded to make this work? Or could we even be so
> lucky that somebody made RPMs for this?

Perhaps if you stated more clearly the actual errors some one might be
able to help you.  What worries me, is you don't seem to know where you
got the source from. 


-- 
Dave Cotton <[EMAIL PROTECTED]>

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[Asterisk-Users] sip show peers

[Samy Antoun]

Hi,

I have 3 SIP extensions, setup as follows:
 #  Device  Location options
200 Sipura  local
210 Sipura  remote   nat=yes qualify=yes
310 eyebeam remote   nat=yes qualify=yes

This is the result of sip show peers:
Name/user Host  Dyn Nat Status
200/200   192.168.1.150  D  Unmonitored
210/210   84.36.36.140   D   N  OK (305 ms)
310/310   71.180.126.60  D   N  Unmonitored

Does anyone know why ext 210 the only one has a ping
status "OK (305 ms)" and the others are "Unmonitored"

Regards




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Re: [Asterisk-Users] ASTCC -- semantic note of 'callstart' in cdrs?

[Darren Wiebe]

This will actually be easy to fix.  I'll post a patch along with 
someother stuff shortly.


Darren

Darren Wiebe wrote:

That is true.  It's just one of those things that is easier to leave 
alone to avoid breakage in upgrades.  It would be nice to get fixed 
though


Darren Wiebe
[EMAIL PROTECTED]

Eric Lyons wrote:

Looking at the code, it would appear that the 'callstart' column of 
the cdrs table should really be called 'callend':


   $dialstr = "IAX2/$res->{path}/$phone|30|HL(" . 
($maxtime * 60 * 1000) . ":6:3)";

   $res = $AGI->exec("DIAL $dialstr");
   $answeredtime = $AGI->get_variable("ANSWEREDTIME");
   $dialstatus = $AGI->get_variable("DIALSTATUS");
   $callstart = localtime();
   return $dialstatus;

No?

Eric.
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Re: [Asterisk-Users] Direct Dial In - second try

[ChB]

Finally(!), the answer is 

in /etc/asterisk/zapata.conf add

'overlapdial=yes'

hope that helps someone with the same problem.

many thanks to [EMAIL PROTECTED] and gerold


On Mon, 3 Oct 2005 23:55:21 +0200
ChB <[EMAIL PROTECTED]> wrote:

> Hi all,
> 
> I have an asterisk-server (cvs-head from august) connected to a
> carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems
> with DDI (standard 'official pstn' number plus extra digits for
> 'internal' use)
> 
> Basically, when the entire number (including the extra digits) is
> dialled via a redial or a programmed key, I see the entire called party
> number (including the extra numbers) in * debug. However if I dial the
> number manually (digit for digit) the carrier puts it through before
> I've  finished dialling (after 10 digits), and I never see the whole DDI
> dialed.
> 
> Since I want to be able to use numbers with and without DDIs, I can't
> tell the carrier to raise the minimum level to more than 10 since
> numbers dialled without DDIs would never be passed.
> 
> The carrier is telling me to wait a second or two before seizing the
> call, and that any additional digits received would be passed in the
> isdn protocol. My understanding is that the called party number would be
> retransmitted, including the DDI. (Wait in the dialplan doesn't work, as
> the call is already "taken")
> 
> Does anyone have any idea why this is not working? Is the carrier right?
> This doesn't seem to be such an odd feature that nobody else would use
> it ;-), so please leave a comment, even if it works for you out of the
> box without fuss. Your help is highly appreciated, thanks!
> 
> I have included two traces. The first is the number dialled with a speed
> dial key, the second is exactly the same number dialed manually. 
> 
> Best regards
> Christian
> 
> 
> trace:
> 
> 
> 
> with speeddial(sent as one block):
> 
> mgw1*CLI>
> < Protocol Discriminator: Q.931 (8)  len=35 < Call Ref: len= 2
> (reference 1549/0x60D) (Originator) < Message type: SETUP (5) < [04 03
> 80 90 a3] < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info
> transfer capability: Speech (0)
> <  Ext: 1  Trans mode/rate: 64kbps,
> circuit-mode (16)
> <  Ext: 1  User information layer 1: A-Law
> (35)
> < [18 03 a1 83 82]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
> Dchan: 0
>  <   Ext: 1  Coding: 0   Number Specified   Channel
> Type: 3
> <   Ext: 1  Channel: 2 ]
> < [6c 02 00 a1]
> < Calling Number (len= 4) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
> Unknown Number Plan (0)
> <   Presentation: Presentation prohibited, user
> number passed network screening (33) '' ]
> < [70 0e a1 30 37 32 30 30 30 33 34 35 36 37 38 39] < Called Number
> (len=16) [ Ext: 1  TON: National Number (2)  NPI: ISDN/Telephony
> Numbering Plan (E.164/E.163) (1) '07xx6789' ]
> -- Making new call for cr 1549
> -- Processing Q.931 Call Setup
> -- Processing IE 4 (cs0, Bearer Capability)
> -- Processing IE 24 (cs0, Channel Identification)
> -- Processing IE 108 (cs0, Calling Party Number)
> -- Processing IE 112 (cs0, Called Party Number)
> > Protocol Discriminator: Q.931 (8)  len=10 Call Ref: len= 2 (reference 
> > 1549/0x60D) (Terminator) Message type: CALL PROCEEDING (2)
> > [18 03 a9 83 82]
> > Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
> Dchan: 0
> >ChanSel: Reserved
> >   Ext: 1  Coding: 0   Number Specified   Channel
> Type: 3
> >   Ext: 1  Channel: 2 ]
> -- Accepting call from '' to '07xx6789' on channel 0/2, span 1
> -- Executing GotoIf("Zap/2-1", "0?100:2") in new stack
> 
> 
> 
> 
> dialed manually digit by digit:
> 
> < Protocol Discriminator: Q.931 (8)  len=32 < Call Ref: len= 2
> (reference 1543/0x607) (Originator) < Message type: SETUP (5) < [04 03
> 80 90 a3] < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info
> transfer capability: Speech (0)
> <  Ext: 1  Trans mode/rate: 64kbps,
> circuit-mode (16)
> <  Ext: 1  User information layer 1: A-Law
> (35)
> < [18 03 a1 83 82]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
> Dchan: 0
>  <   Ext: 1  Coding: 0   Number Specified   Channel
> Type: 3
> <   Ext: 1  Channel: 2 ]
> < [6c 02 00 a1]
> < Calling Number (len= 4) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
> Unknown Number Plan (0)
> <   Presentation: Presentation prohibited, user
> number passed network screening (33) '' ]
> < [70 0b a1 30 37 32 30 30 30 33 34

Re: [Asterisk-Users] Quad BRI with Fedora, anyone?

[Mir]

Actually, the QuadBRI card is not from Digium, but manufactured by Junghanns.net

Michael

2005/10/15, Lars Dybdahl <[EMAIL PROTECTED]>:
> We have a QuadBRI ISDN card from Digium. We would like to make it work
> with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of
> bristuff from the Digium homepage fails, both the stable version with
> asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD.
>
> Has anybody here succeeded to make this work? Or could we even be so
> lucky that somebody made RPMs for this?
>
> Lars Dybdahl.
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[Asterisk-Users] Planet Vip-150T

[FaberK]

Hi All,
I'm having problem with this phone.
Problems are regarding voicemail message alert on the phone.
---
handle_response: Host 'xxx.xxx.xxx.xxx' does not implement 'NOTIFY'
---
Can somebody help?
On the phone manual, is written that it can acept MWI, but... not mine!!!

Thanks!
--
.:FaberK:.
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Re: [Asterisk-Users] Hints and Call Waiting

[Tom Rymes]

We use Cisco phones and we simply disabled call-waiting for those  
lines. Don't know if that will help, but whatever soft/hardphone you  
are using probably has a way to disable call-waiting.


Tom

On Oct 15, 2005, at 5:38 AM, João Paulo Antunes wrote:


Hi!

We have a big problem in our call center: when an agent does an  
outgoing call it can receive calls from the queues. The same  
happens if one agent transfer a call for another agent... and the  
ringing tone while in a call is puting the agents like crazy...


We have the hints working with lines like this in extensions.conf:

exten => 101,hint,SIP/101

If we set incominglimit to 1 the agent cannot do another call (to  
do attended transfers)


We are using Beta1

Can anyone help?

Thanks,
Joao Antunes


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Re: [Asterisk-Users] Attended Call Transfer

[Tom Rymes]

Search voip-info.org or google for features.conf.TomOn Oct 15, 2005, at 4:58 AM, Denis Vella wrote: Hi,   We're trying to setup attended call transfer, but we have not been able to find the required configuration.   Blind transfer works fine using the # key, but we don't like the fact that the transferring extension does not have any info on what happened to the call.   Any pointers to setting up proper attended transfer?   Thanks, Denis The information contained in this email is confidential and may be privileged. It is intended for the addressee only, if you are not the intended recipient please notify the sender and delete the email immediately. The contents of this email must not be disclosed or copied without the senders consent. We cannot accept any responsibility for viruses. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Philip Toledo Limited___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [Asterisk-Users] Problem with '#' key recognition

[Colin Martin]

Tzafrir Cohen wrote:

On Sat, Oct 15, 2005 at 03:21:38PM +0100, Colin Martin wrote:


Hi,

I seem to be unable to get Asterisk to recognise the '#' key being 
pressed to acknowledge an incoming call from a queue. No matter how many 
times I press the key to acknowledge, the Asterisk server acts as if I 
have not.


I have installed the ztdummy module, and it seems that Asterisk is 
picking it up OK (mainly since it's not moaning about permissions for 
IAX timing any more).


I am using the Asterisk version that is packaged in Debian Stable 
(Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k) and the call that goes out 
expecting a '#' goes over an IAX provider to a UK mobile/landline.



What phone do you use? How is it connected to Asterisk?



I have tried all the following phones:

Cisco 7940 (via SIP)
UK Mobile, Orange, Samsung D500 (via IAX to outbound provider)
UK Landline (via IAX to outbound provider)

It just seems that Asterisk totally ignores the '#' key.

I found the following on
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin

"(Note: Without a valid zaptel timing source, the '#' acknowledgement 
will not happen.) "


Although I have installed the ztdummy module, is there any way to get 
Asterisk to tell me if it is working or not?


Many thanks for the response.

Colin


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Re: [Asterisk-Users] Problem with '#' key recognition

[Tzafrir Cohen]

On Sat, Oct 15, 2005 at 03:21:38PM +0100, Colin Martin wrote:
> Hi,
> 
> I seem to be unable to get Asterisk to recognise the '#' key being 
> pressed to acknowledge an incoming call from a queue. No matter how many 
> times I press the key to acknowledge, the Asterisk server acts as if I 
> have not.
> 
> I have installed the ztdummy module, and it seems that Asterisk is 
> picking it up OK (mainly since it's not moaning about permissions for 
> IAX timing any more).
> 
> I am using the Asterisk version that is packaged in Debian Stable 
> (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k) and the call that goes out 
> expecting a '#' goes over an IAX provider to a UK mobile/landline.

What phone do you use? How is it connected to Asterisk?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] AGI Variable problem

[Moises Silva]

kind of difficutl to help you if you dont provide the script or
relevant info about your php.ini configuration, stuff related to the
output buffer.

I used to do programming with phpagi class, but then i came up with something more simple and usefull for my purposes. 

http://galileo.ivsol.net/scripts/AgiPhp5.php

only works with php, but with small modifications can work with php4.

best regardsOn 10/13/05, René Enskat [Teamware GmbH] <[EMAIL PROTECTED]> wrote:
The ("/var/lib/asterisk/agi-bin/phpagi.php") is the newest form the siteupdated today, and i wrote the script like other examples and i can't find asyntax mistake inside extension.conf and the php script .(
On Thu, 13 Oct 2005 09:32:14 -0500  Moises Silva <[EMAIL PROTECTED]> wrote:> for some reason your script is not executing the get_var correctly, as you
> can see in the output, asterisk is saying: "invalid or unknown command".>> check the internals of your script, the most common reason is that you are> mispelling the command.>
> best regards>> On 10/13/05, René Enskat [Teamware GmbH] <[EMAIL PROTECTED]> wrote:>> Hello all, I try to use a agi script to get a variable from * und put them into a
>> script which gives me another variablke and put this in *.>> My problem is now it seems the var ID is empty coz i always jump into>> the result 0 loop.>> The $MSN should be in the SetCIDNum.
 #!/usr/bin/php -q >> include("/var/lib/asterisk/agi-bin/phpagi.php");>> $agi = new AGI(); $ID = $agi->get_variable("SIPUSER");
 if ($ID['result'] == 0) {>> $agi->verbose("SIPUSER not set -- nothing to do");>> exit(1);>> } $agi->set_variable("MSN", exec("/var/lib/asterisk/agi-bin/msn4sip 111
>> 222 333 " .$ID['data']));>> ?> Output from asterisk:>> -- Executing SetVar("SIP/31-79e2", "SIPUSER=31") in new stack>> -- Executing AGI("SIP/31-79e2", "
msn4sip.agi") in new stack>> -- Launched AGI Script /var/lib/asterisk/agi-bin/msn4sip.agi>> msn4sip.agi: Arrayn(n [code] => 510n [result] => n [data] =>>> Invalid or unknown commandn)n
>> msn4sip.agi: SIPUSER not set -- nothing to do>> -- AGI Script msn4sip.agi completed, returning 0>> -- Executing SetLanguage("SIP/31-79e2", "de") in new stack>> -- Executing SetCIDNum("SIP/31-79e2", "") in new stack
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Re: [Asterisk-Users] Maintenance panel

[Tom Vile]

thats an Asterisk At Home mod.On 10/15/05, Tommy Denton <[EMAIL PROTECTED]> wrote:
Where can I get the maintenance panel for AMP?  I have searched all over and cannot seem to find it.

Thank you in advance,

Tommy

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http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856
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[Asterisk-Users] Maintenance panel

[Tommy Denton]

Where can I get the maintenance panel for AMP?  I have searched all over and cannot seem to find it.

Thank you in advance,

Tommy
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[Asterisk-Users] Problem with '#' key recognition

[Colin Martin]

Hi,

I seem to be unable to get Asterisk to recognise the '#' key being 
pressed to acknowledge an incoming call from a queue. No matter how many 
times I press the key to acknowledge, the Asterisk server acts as if I 
have not.


I have installed the ztdummy module, and it seems that Asterisk is 
picking it up OK (mainly since it's not moaning about permissions for 
IAX timing any more).


I am using the Asterisk version that is packaged in Debian Stable 
(Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k) and the call that goes out 
expecting a '#' goes over an IAX provider to a UK mobile/landline.


I have Googled on the subject but can't seem to find anything useful. I 
was wondering if anyone had any ideas or places to look before I start 
tearing my hair out.


Thanks,

Colin
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Re: [Asterisk-Users] Busy not jumping n + 101 anymore

[Ronald Wiplinger]

Kevin P. Fleming wrote:


Eric "ManxPower" Wieling wrote:


Sorry, it's in asterisk/configs/extensions.conf.sample



And the default is supposed to be 'on', so that it is backwards 
compatible unless you turn it off (which is in the sample config file 
so that new users will learn to build their dialplans with it turned 
off). If it is defaulting to "off" for some reason, that is a bug.

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When exactly was THAT change? I upgraded recently, but did not see that 
behaviour:


Asterisk CVS HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 
2005-10-11 01:01:53 UTC



bye

Ronald Wiplinger

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Re: [Asterisk-Users] Busy not jumping n + 101 anymore

[Kevin P. Fleming]

Eric "ManxPower" Wieling wrote:


Sorry, it's in asterisk/configs/extensions.conf.sample


And the default is supposed to be 'on', so that it is backwards 
compatible unless you turn it off (which is in the sample config file so 
that new users will learn to build their dialplans with it turned off). 
If it is defaulting to "off" for some reason, that is a bug.

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[Asterisk-Users] Quad BRI with Fedora, anyone?

[Lars Dybdahl]

We have a QuadBRI ISDN card from Digium. We would like to make it work
with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of
bristuff from the Digium homepage fails, both the stable version with
asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD.

Has anybody here succeeded to make this work? Or could we even be so
lucky that somebody made RPMs for this?

Lars Dybdahl.
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[Asterisk-Users] Hints and Call Waiting

[João Paulo Antunes]

Hi!

We have a big problem in our call center: when an agent does an outgoing 
call it can receive calls from the queues. The same happens if one agent 
transfer a call for another agent... and the ringing tone while in a 
call is puting the agents like crazy...


We have the hints working with lines like this in extensions.conf:

exten => 101,hint,SIP/101

If we set incominglimit to 1 the agent cannot do another call (to do 
attended transfers)


We are using Beta1

Can anyone help?

Thanks,
Joao Antunes


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[Asterisk-Users] Attended Call Transfer

[Denis Vella]

Title: Message



Hi,
 
We're trying to 
setup attended call transfer, but we have not been able to find the 
required configuration.
 
Blind transfer works 
fine using the # key, but we don't like the fact that the transferring 
extension does not have any info on what happened to the 
call.
 
Any pointers to 
setting up proper attended transfer?
 
Thanks,
Denis The information contained in this email is confidential and may be privileged. It is intended for the addressee only, if you are not the intended recipient please notify the sender and delete the email immediately. The contents of this email must not be disclosed or copied without the senders consent. We cannot accept any responsibility for viruses. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Philip Toledo Limited
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[Asterisk-Users] Disconnecting after 1 min while Communicating Clarent class 5 call manager

[Anil Kumar K]

Hi List
I installed asterisk server and tried to transfer calls from asterisk to Clarent class 5 call manager.
The calls are passing through with out any problem but after 60 seconds the call get disconnected automatically. 
Please help me to sort out this problem. Attaching here with my sip configuration file

Sip.conf.
[general]
rtpholdtimeout=300
rtptimeout=300
defaultexpirey=20
context=default     
port=5060 
 
bindaddr=0.0.0.0    
srvlookup=yes    

[clarent]
register => test:[EMAIL PROTECTED]/123456
type=friend
secret=1234
username=testuser
host=192.168.10.150
fromuser=test1234
insecure=very
fromdomain=192.168.10.150
canreinvite=no
nat=no
disallow=all
allow=g723
context=default


Thanks in advance.

Anil
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Re: [Asterisk-Users] Busy not jumping n + 101 anymore

[Eric \"ManxPower\" Wieling]

Andrew Kohlsmith wrote:

On Friday 14 October 2005 20:50, Eric "ManxPower" Wieling wrote:


Oddly enough, I believe it's mentioned in UPGRADE.txt.



Care to tell us where?  I just checked my CVS HEAD copy of UPGRADE.txt.


Sorry, it's in asterisk/configs/extensions.conf.sample
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