Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)

[Lubomir Christov]

Jeremy McNamara wrote:
Lubomir Christov wrote:

BUT I have the say that I have the same opinion as martin 
([EMAIL PROTECTED]): "Although personally I would prefer oh323 for 
its very well described config file for now winner is chan_h323"


Again, what is not clear about h323.conf?  It follows the other Asterisk 
channel driver config methods.

I am looking for specifics here, not just generalizations. 

There are some missing useful features in chan_h323 (which are present 
in chan_oh323)

1. missing jitter buffer - I know that this is because of the asterisk 
rtp limits, It don't have such a feature

2. LogLevel/LogFile feature in config - right now (as I know) if you 
don't start asterisk on console (asterisk -c) you can't receive the 
trace output from chan_h323 and it is impossible to redirect the trace 
output to some file from h323.conf (like 
libLogFile=/var/log/asterisk/oh323 is doing it)

Lubo

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[Asterisk-Users] SIMPLE support in Asterisk?

[Kerker Staffan]

Hi
Is there any work being done on implementing IM/SIMPLE support
for SIP on Asterisk? Like a presence server? 

rdgs,
/Staffan Kerker

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[Asterisk-Users] For all IAXTEL users of DIAX

[Dan]

Hi all.

In order to dial an IAXTEL number, even you have registered using
user/pass/iaxtel.com, at the end of the dial string a @iaxtel context must
be appended.
I will do this automatically in version 0.9.5
In the mean time, you can only dial IAXTEL numbers from the phonebook.

Sorry for the inconvenience.

Best regards,
Dan
P.S. If someone can send me his IAXTEL number, I'll call him to test this
and to see my own IAXTELnumber (because I have a registration, but I forgot
my number..:))

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[Asterisk-Users] what is the problem?

[C M]

can u guys see what is the problem here. i am a
newbee:D and i want help.

my conf files for reference:

...iax.conf

[general]
register => hsbohra:[EMAIL PROTECTED]

[NuFone]
type=user
secret=mysecret
context=nufone1

[NuFonePeer]
type=peer
secret=mysecret
context=usacall
host=switch-1.nufone.net

extensions.conf

[usacall]
exten=>_1NXXN,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}

[nufone1]
exten=>8002488870,1,Answer

where is [NuFone] used? veryfying from nufones server?

cm

=
Designs

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Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)

[Max Tulyev]

В сообщении от 24 Ноябрь 2003 21:27 Jeremy McNamara написал:

> I would like to hear from anyone else that has real world experiences
> with both chan_h323 and asterisk-oh323.

I have asterisk-oh323-0.5.7.tar.gz and * from CVS @ 20 Nov 2003.
PWLib 1.5.2, OpenH323 1.12.2

ATA-186(h.323)->gnugk->*->7940(SIP)

I see segmentation fault on asterisk when RTP starts. I don't know why, I'll 
try to find it. Now chan_h323 works fine instead of OH323.


-- 
С Уважением,
Максим Тульев (MT6561-RIPE, 2:463/[EMAIL PROTECTED])
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[Asterisk-Users] cdr_unixodbc

[Brian West]

asterisk*CLI> load cdr_unixodbc.so
 Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR Backend)
  == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing 
'/etc/asterisk/cdr_unixodbc.conf': Found
-- cdr_unixodbc: dsn is MySQL-asterisk
-- cdr_unixodbc: username is root
-- cdr_unixodbc: password is [secret]
-- Connected to MySQL-asterisk

it works.. it logs calls.  Anyone else intrested?  For now I have to do
more clean up... ya know make it pretty. :)

bkw
PS unixODBC is LGPL :)
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Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)

[Michael Manousos]

Adam Hart wrote:
From: "Jeremy McNamara" <[EMAIL PROTECTED]>

I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.
Be brutal.  I want to know the gory details, so we can stop any future
pissing matches from even starting by having everything publicly
documented for all newbies.
Jeremy McNamara


I've used both, I find chan_h323 more robust as it does less. Given it uses
*'s rtp, means less chance of errors but also inherits problems of rtp.c
(inability to set payload) My chan_oh323 still crashes on exit which
concerns me on the quality. (I don't really care if it's on exit). Having
So lets fix it!
I use asterisk-oh323 without problems for my setup.
If you provide the needed information I can fix it.
said that, the config options in chan_oh323 are great - the ability to turn
on and off fast start and change the packet payload sizes is handy. My main
problem with chan_h323 was G.729, which caused me switching to chan_oh323
I'm using chan_oh323 to pass through many G.729 calls to/from
Cisco 17xx boxes without a single problem.
but given my patch has been added, that problem is solved. Speaking of
patches, an unknown feature I submitted was nat=true (same as sip.conf) for
chan_h323. I haven't heard anyone using this so let me know if it does or
doesn't work.
-Adam



Michael.



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Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)

[Michael Manousos]

Max Tulyev wrote:
В сообщении от 24 Ноябрь 2003 21:27 Jeremy McNamara написал:


I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.


I have asterisk-oh323-0.5.7.tar.gz and * from CVS @ 20 Nov 2003.
PWLib 1.5.2, OpenH323 1.12.2
ATA-186(h.323)->gnugk->*->7940(SIP)

I see segmentation fault on asterisk when RTP starts. I don't know why, I'll 
If you don't provide the needed info the problem
will remain.
try to find it. Now chan_h323 works fine instead of OH323.




Michael.

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Re: [Asterisk-Users] Strange code in rtp.c / disconnect - maybe reinvite problems

[Detlef Wengorz]

Daniel Chabrol wrote:
> 
> Hi List!
> 
> I get "WARNING[14351]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 =
> 524300 is not codec1 = 524300, can't do reinvite" at my asterisk console.
> 
> The code there looks realy strange:
> 
> codec0 = pr0->get_codec(c0);
> codec1 = pr1->get_codec(c1);
> ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, can't do
> reinvite\n",codec0,codec1);
> /* Hey, we can't do reinvite if both parties speak diffrent codecs */
> if (codec0 != codec1)
> return -2;
> 
> I think the message should only occur *after* checking equality:
> 
> if (codec0 != codec1) {
> ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, can't do
> reinvite\n",codec0,codec1);
> return -2;
> }
> 
> I hoped this "can't do reinvite" would explain my disconnects from the
> nikotel.com sip server after 60 seconds. But this little bug seems only
> to be display-specific and not affect funtion. But maybe i oversight

That's correct :-(
but change the code like this

if (codec0 != codec1) {
 ast_log(LOG_WARNING,
 "codec0 = %d is not codec1 = %d, can't do
reinvite\n",codec0,codec1);

 ast_mutex_unlock(&c0->lock); // unlock before return
 ast_mutex_unlock(&c1->lock); // unlock before return
 return -2;
}

and try again.
maybe it helps.




> something which still disables the reinvite even if i use
> canreinvite=yes in my sip.conf:
> 
> [nikotel]
> type=friend
> username=USERID
> fromuser=USERID
> secret=PASSWORD
> host=calamar0.nikotel.com
> canreinvite=yes
> context=internal
> ; no nat entry because im not using nat!
> 
> Is there someone which is able to use Nikotel.com with the current
> CVS-Version (in my case CVS-11/24/03-19:24:22). BTW: 0.5.0 don't work
> too in my case (at least not longer than 60 seconds). Pulver.com calls
> and so on are no problem. Any suggestions?
> 
> Best regards,
> Daniel
> 
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Detlef Wengorz <[EMAIL PROTECTED]>
Detlef Wengorz <[EMAIL PROTECTED]>
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Re: [Asterisk-Users] ISDN Card Types for Europe

[Roy Sigurd Karlsbakk]

I don't know if they're finished yet, but Klaus-Peter Junghanns is
working on some quad BRI passive HFC-PCI cards that will probably
'killall -9 avm diva' both in functionality (native drivers for
asterisk) and in price.

roy

On Tue, 2003-11-18 at 17:01, Ray Burkholder wrote:
> What types of ISDN BRI cards work well in Europe (Guadeloupe,
> Martinique and France) ?  For example:  AVM C2 or AVM C4 or eicon Diva
> server 4 BRI?  Any others?  Which driver is appropriate?
> 
> Ray Burkholder
> [EMAIL PROTECTED]
> http://www.oneunified.net
> 704 576 5101
> 
> 
> -- 
> Scanned for viruses & dangerous content at One Unified and is believed
> to be clean.

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RE: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download

[Florian Overkamp]

Hi, 

> -Original Message-
> > There is no sound on either side of the call, so I guess that
> qualifies as
> > 'it
> > doesnt work for me' :-P
> 
> Do you have:
> 
> Disallow=All
> Allow=GSM
> 
> 
> In the section of iax.conf for the user using DIAX ?

Hey, you're right, that helps! But I don't understand, taking a peek at the
iax2 channel status shows it _is_ infact using GSM, so I don't understand
why it should make a difference :-(

Thanks for the suggestion - this is a workaround as far as I'm concerned!

Best regards,
Florian

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Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)

[Roy Sigurd Karlsbakk]

If it was possible to get any support at all from Jeremy (or others),
I'd be glad to use it. I have sent numerous reports with where it failed
and what I did to remedy this without getting any response. With
developers this arrogant, chan_h323 should be removed from the asterisk
tree, unless the code support is improved.

roy

On Mon, 2003-11-24 at 19:27, Jeremy McNamara wrote:
> I would like to hear from anyone else that has real world experiences 
> with both chan_h323 and asterisk-oh323.
> 
> Be brutal.  I want to know the gory details, so we can stop any future 
> pissing matches from even starting by having everything publicly 
> documented for all newbies.
> 
> 
> 
> Jeremy McNamara
> 
> 
> 
> 
> 
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[Asterisk-Users] * Configuration

[Girish Gopinath]

Hi,

I am a beginner to Asterisk. Can anybody clear my following doubts regarding
the configuration needed?
1) What is the ideal system configuratin required?(like processer, RAM, h/d 
space etc)
2) How many connections it can handle at a time?
3) How many Virtual PBXs it can handle?
4) Whether Postgres or Mysql is best suited?
5) How many IVR's it can handle simultaneously?
6) How many Voicemails can be recorded at a time?
7) What type of bandwidth does * require?

Thanx and Regards...

Girish Gopinath

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Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)

[Michael Manousos]

Jeremy McNamara wrote:


As history shows I was totally blown off by Michael when I offered to 
help better his driver.  Then I was even told that I couldn't create 
anything better...hence the birth of chan_h323 and this whole mess.
Yes, sure, whatever.



Jeremy McNamara

Michael.

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Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download

[Michael Bielicki]

check both directions
when you do a show channel ... does it show gsm in both ways ?
Florian Overkamp wrote:
Hi, 

 

-Original Message-
   

There is no sound on either side of the call, so I guess that
 

qualifies as
   

'it
doesnt work for me' :-P
 

Do you have:

Disallow=All
Allow=GSM
In the section of iax.conf for the user using DIAX ?
   

Hey, you're right, that helps! But I don't understand, taking a peek at the
iax2 channel status shows it _is_ infact using GSM, so I don't understand
why it should make a difference :-(
Thanks for the suggestion - this is a workaround as far as I'm concerned!

Best regards,
Florian
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Re: [Asterisk-Users] * Configuration

[wasim]

On Tue, 25 Nov 2003, Girish Gopinath wrote:

> 1) What is the ideal system configuratin required?(like processer, RAM, h/d 
> space etc)

depends (on what codec, what users, whats the purpose)

> 2) How many connections it can handle at a time?

depends on 1 above

> 3) How many Virtual PBXs it can handle?

depends on what a virtual pbx is expected to do, theoretical limit?

> 4) Whether Postgres or Mysql is best suited?

depends, cdr_mysql is in addons, bkw_ just did unixodbc, what do you 
prefer?

> 5) How many IVR's it can handle simultaneously?

depends on 1

> 6) How many Voicemails can be recorded at a time?

depends (on codec, available i/o)

> 7) What type of bandwidth does * require?

depends (on codec and protocol)

> Thanx and Regards...

most welcome..
i was just going to write depends to all, but you get the point ...
best go through the archives in detail, ALL your answers lie in that...

- wasim
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[Asterisk-Users] How to use * to simply skim off callerid (UK)?

[Dave Wilson]

Hi all,

I'm wondering how I could go about placing * into an existing office PBX
system so as to capture callerid for further processing via AGI into an
intranet app.

I've already got the AGI scripts set up for what I want to do and have
tested using IAX2 peering, however I have no knowledge of the target
environment as yet, so would like to know what do I need to look out for
(line types, hardware, cabling, etc)? My company builds automotive dealer
management systems and we wish to use * within the system suite but without
forcing any changes to potential client's existing phone systems in the
initial install. Our primary concern is that of garnering the callerid info
so as to trigger screen prompts and other events within our own system.

Can anyone tell me what PBX systems on the UK market it is currently
possible to connect to * with? Once * and our system is in place within a
client network, we shall then be pushing for full leverage of *'s features.

TIA,
Dave


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Re: [Asterisk-Users] * Configuration

[Girish Gopinath]

From: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: Girish Gopinath <[EMAIL PROTECTED]>
CC: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] * Configuration
Date: Tue, 25 Nov 2003 15:24:11 +0500 (PKT)
On Tue, 25 Nov 2003, Girish Gopinath wrote:

> 1) What is the ideal system configuratin required?(like processer, RAM, 
h/d
> space etc)

depends (on what codec, what users, whats the purpose)

> 2) How many connections it can handle at a time?

depends on 1 above

> 3) How many Virtual PBXs it can handle?

depends on what a virtual pbx is expected to do, theoretical limit?

> 4) Whether Postgres or Mysql is best suited?

depends, cdr_mysql is in addons, bkw_ just did unixodbc, what do you
prefer?
> 5) How many IVR's it can handle simultaneously?

depends on 1

> 6) How many Voicemails can be recorded at a time?

depends (on codec, available i/o)

> 7) What type of bandwidth does * require?

depends (on codec and protocol)

> Thanx and Regards...

most welcome..
i was just going to write depends to all, but you get the point ...
best go through the archives in detail, ALL your answers lie in that...
- wasim
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[Asterisk-Users] Ring requested on channel 1 already in use...

[Alastair Maw]

I'm running an E400P. Every now and then Asterisk stops receiving 
incoming calls.



This turns up in the messages log:

Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793 
(pri_dchannel): Ring requested on channel 1 already in use on span 1. 
Hanging up owner.

Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793 
(pri_dchannel): Ring requested on channel 1 already in use on span 2. 
Hanging up owner.

Nov 25 10:49:25 WARNING[98311]: File chan_zap.c, Line 5793 
(pri_dchannel): Ring requested on channel 1 already in use on span 3. 
Hanging up owner.

Nov 25 10:49:25 WARNING[114696]: File chan_zap.c, Line 5793 
(pri_dchannel): Ring requested on channel 1 already in use on span 4. 
Hanging up owner.



A little while back I also had this in my logs:

Nov 15 17:25:21 WARNING[114696]: File chan_zap.c, Line 5790 
(pri_dchannel): Duplicate setup requested on channel 11 already in use 
on span 4

Nov 15 17:25:21 WARNING[65541]: File chan_zap.c, Line 5790 
(pri_dchannel): Duplicate setup requested on channel 4 already in use on 
span 1

Nov 15 17:25:21 WARNING[114696]: File chan_zap.c, Line 5790 
(pri_dchannel): Duplicate setup requested on channel 12 already in use 
on span 4

Nov 15 17:25:21 WARNING[65541]: File chan_zap.c, Line 5790 
(pri_dchannel): Duplicate setup requested on channel 5 already in use on 
span 1

Nov 15 17:25:22 WARNING[65541]: File chan_zap.c, Line 5790 
(pri_dchannel): Duplicate setup requested on channel 3 already in use on 
span 1

Nov 15 17:25:22 WARNING[65541]: File chan_zap.c, Line 5790 
(pri_dchannel): Duplicate setup requested on channel 2 already in use on 
span 1

Nov 15 17:25:24 WARNING[114696]: File chan_zap.c, Line 5790 
(pri_dchannel): Duplicate setup requested on channel 13 already in use 
on span 4

FWIW, my libpri/zaptel/asterisk installs are all about two months old. 
Might whatever causes this have been fixed by now? (I don't want to 
upgrade otherwise as this problem is quite intermittent).

Anyone have any ideas?

Alastair

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[Asterisk-Users] Problem with fax detection

[Micke Andersson]

Hi

I have a problem with the fax detection.

I want to be able to turn that of on all zap channels.

the * is in between my E1 and my PBX and when I try to make a fax call out
on the E1 the * detects the fax tone and hangsup the outgoing zap channel.

How should I solve this ?

/Mike

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RE: [Asterisk-Users] cdr_unixodbc

[Asterisk]

Brian!

You've done something tricky again! I'm interested! *wink*

Where do I get it?

;)
Ben


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Ph: (+61) 1300 665 575
Fx: (+61) 1300 556 595



-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED] 
Posted At: Tuesday, 25 November 2003 7:28 PM
Posted To: Asterisk
Conversation: [Asterisk-Users] cdr_unixodbc
Subject: [Asterisk-Users] cdr_unixodbc


asterisk*CLI> load cdr_unixodbc.so
 Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR
Backend)
  == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing
'/etc/asterisk/cdr_unixodbc.conf': Found
-- cdr_unixodbc: dsn is MySQL-asterisk
-- cdr_unixodbc: username is root
-- cdr_unixodbc: password is [secret]
-- Connected to MySQL-asterisk

it works.. it logs calls.  Anyone else intrested?  For now I have to do
more clean up... ya know make it pretty. :)

bkw
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Re: [Asterisk-Users] cdr_unixodbc

[Pavel Litvinenko]

Brian West wrote:

asterisk*CLI> load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR Backend)
 == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing 
'/etc/asterisk/cdr_unixodbc.conf': Found
   -- cdr_unixodbc: dsn is MySQL-asterisk
   -- cdr_unixodbc: username is root
   -- cdr_unixodbc: password is [secret]
   -- Connected to MySQL-asterisk
it works.. it logs calls.  Anyone else intrested?  For now I have to do
more clean up... ya know make it pretty. :)
bkw
PS unixODBC is LGPL :)
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Yes,  I do :) where can I get it  ?

--

-
Best Regards,
Pavel Litvinenko.
ICQ: 16224754
Ph: (8632) 923962, 923640
sip:[EMAIL PROTECTED]


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Re: [Asterisk-Users] cdr_unixodbc

[WipeOut]

Pavel Litvinenko wrote:

Brian West wrote:

asterisk*CLI> load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR 
Backend)
 == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing 
'/etc/asterisk/cdr_unixodbc.conf': Found
   -- cdr_unixodbc: dsn is MySQL-asterisk
   -- cdr_unixodbc: username is root
   -- cdr_unixodbc: password is [secret]
   -- Connected to MySQL-asterisk

it works.. it logs calls.  Anyone else intrested?  For now I have to do
more clean up... ya know make it pretty. :)

Yes,  I do :) where can I get it  ?

What are the software requirements?

I have never used unixodbc before.. :)

Later..

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RE: [Asterisk-Users] cdr_unixodbc

[Adams, Gavin]

> -Original Message-
> From: Brian West [mailto:[EMAIL PROTECTED]
> 
> asterisk*CLI> load cdr_unixodbc.so
>  Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR
> Backend)
>   == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing
> '/etc/asterisk/cdr_unixodbc.conf': Found
> -- cdr_unixodbc: dsn is MySQL-asterisk
> -- cdr_unixodbc: username is root
> -- cdr_unixodbc: password is [secret]
> -- Connected to MySQL-asterisk
> 
> it works.. it logs calls.  Anyone else intrested?  For now I have to
do
> more clean up... ya know make it pretty. :)

Good stuff _bkw! Anything that abstracts out the database connectivity
is a good thing. Opens up the world to a host of databases! Makes *
palatable for the non-MySQL/PostgreSQL crowd.

Regards,

--- Gavin

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Re: [Asterisk-Users] cdr_unixodbc

[Brian West]

Ok the basic requirement is unixODBC and the MyODBC driver(for MySQL) or
you can pick what ever you want(http://www.unixodbc.org/drivers.html).
The table structure is the same as pgsql or mysql ... just duplicate that.

I would like to verify that I have done this in such a way that the
database you choose to use won't matter(unixODBC is all about that)

odbc.ini info:

[ODBC Data Sources]
MySQL-asterisk = MySQL ODBC Driver Asterisk DSN

[MySQL-asterisk]
Description = MySQL ODBC Driver Asterisk DSN
Driver  = MySQL
Socket  = /var/run/mysqld/mysqld.sock
Server  = localhost
User= root
Database= asterisk
Option  = 3
#Port   =
Password= password

odbcinst.ini

[MySQL]
Description = MySQL ODBC MyODBC Driver
Driver  = /usr/lib/libmyodbc3.so

...  I will give you guys time to do that ... and i'm still cleaning up
the code a bit more.  unixODBC is a bit more forgiving than the MySQL C
API is.


bkw


On Tue, 25 Nov 2003, WipeOut wrote:

> Pavel Litvinenko wrote:
>
> > Brian West wrote:
> >
> >> asterisk*CLI> load cdr_unixodbc.so
> >> Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR
> >> Backend)
> >>  == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing
> >> '/etc/asterisk/cdr_unixodbc.conf': Found
> >>-- cdr_unixodbc: dsn is MySQL-asterisk
> >>-- cdr_unixodbc: username is root
> >>-- cdr_unixodbc: password is [secret]
> >>-- Connected to MySQL-asterisk
> >>
> >> it works.. it logs calls.  Anyone else intrested?  For now I have to do
> >> more clean up... ya know make it pretty. :)
> >>
> >>
> > Yes,  I do :) where can I get it  ?
> >
> What are the software requirements?
>
> I have never used unixodbc before.. :)
>
> Later..
>
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Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download

[firedude]

Dan
I seem to be having the same problem as some of the other guys.  With all 
the previous versions I could make outgoing and receive incoming calls; 
however with this latest version even if I have Diax open the call drops 
through to the busy priority in my extensions.conf file.  It's like it's 
showing Diax is already on a call but this is not the case.  If you have 
some suggestions I'll play around with it a bit but something just doesn't 
seem right with it.  
AJ


On Tue, 25 Nov 2003, Dan wrote:

> Hi Steven,
> 
> - Original Message - 
> From: "Steven Sokol" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, November 25, 2003 4:09 AM
> Subject: RE: [Asterisk-Users] New DIAX - version 0.9.4 - a big step
> forward - available for download
> 
> 
> > > > Dan,
> > > >
> > > > I have been working with the new version and have discovered a
> > strange
> > > > issue.  When I place a call from a DIAX phone, all seems to
> > generally
> > > > work properly.  However, calls placed from either of my SIP agents
> > > > (Grandstream 101 hardphone and X-PRO softphone) or from the PSTN via
> > > > X100P often fail to connect.
> > >
> > > I had a similar experience when I upgraded to 0.9.4.  i was running a
> > cvs
> > > version from late october. blew away my src tree and cvs checkout
> > resolved
> > > it.  Maybe something related to IAX2.
> > >
> > > HTH,
> > > Andy
> >
> > I thought about that -- I did an upgrade from CVS and the issue remains.
> > I haven't gone all the way to a blow-out and re-create.  If anybody else
> > has had a similar experience and that fixed it, please say so.  I will
> > kill /usr/src/asterisk (which is only about a week old) and try again.
> 
> It works for me with:
> Asterisk CVS-11/06/03-10:19:25,
> 
> Best regards,
> Dan
> 
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Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls

[Rich Adamson]

> >From a post to this very thread just a few hours ago:
> 
> > exten => _9.,1,ChanIsAvail(Zap/1&Zap/2)
> > exten => _9.,2,Dial(${AVAILCHAN})
> > exten => _9.,102,NoOp
> > exten => _9.,103,Congestion 

Kind of a side question... what does ChanIsAvail actually check in the
above (loop attached, dialtone, or just that a call is/isn't in progress)?



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[Asterisk-Users] zt_rec: Unknown error 500

[Michiel Betel]

I have a number of Zap/ extensions defined in a queue with ringall 
strategy. When this queue is called sometimes Asterisk seems to think 
that one of these channels is busy, while it is NOT. The following is 
shown on the console:
--Called 44
   -- Called 36
   -- Called 41
   -- Called 35
   -- Called 38
   -- Zap/44-1 is ringing
   -- Zap/36-1 is ringing
   -- Zap/41-1 is ringing
   -- Zap/35-1 is ringing
   -- Zap/38-2 is ringing
   -- Zap/44-1 is ringing
   -- Zap/36-1 is ringing
   -- Hungup 'Zap/35-1'
   -- Zap/41-1 is ringing
   -- Zap/44-1 is ringing
   -- Zap/36-1 is ringing
While a
WARNING[165916]: File chan_zap.c, Line 3331 (zt_read): zt_rec: Unknown 
error 500
is generated in /var/log/asterisk/messages
Any ideas on how to fix this?? Thanks!

Michiel



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Re: [Asterisk-Users] NTT FSK - Japanese Caller ID

[Steve Underwood]

Isamar Maia wrote:

Why don't you put it on the -dev list. Even if most of us might not be
able to help much, we could watch and learn.
   

The issue to create several lists was already decided and implemented?
If so, let me know since I didn't get this thread.
 

I implemented routines to analyse the NTT caller ID format recently, 
because someone else is interested in seeing this work. However, I 
haven't had a chance to test them yet (too busy with other things). The 
code is in my spandsp library, along with my software FAX modem. You can 
find it at ftp://ftp.opencall.org It isn't integrated with * yet, though.

Regards,
Steve
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RE: [Asterisk-Users] cdr_unixodbc

[Vledder, Hans]

Hi Brian,

Excellent job, but how about calling the application 'cdr_odbc' instead of
'cdr_unixodbc', because up to now 'unix' is obvious/trivial when it comes to
* isn't it? Besides, I think 'cdr_odbc' is more in line with cdr_mysql and
cdr_csv and what have you ...

Keep it up !
Hans

-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 9:28 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cdr_unixodbc


asterisk*CLI> load cdr_unixodbc.so
 Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR Backend)
  == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing
'/etc/asterisk/cdr_unixodbc.conf': Found
-- cdr_unixodbc: dsn is MySQL-asterisk
-- cdr_unixodbc: username is root
-- cdr_unixodbc: password is [secret]
-- Connected to MySQL-asterisk

it works.. it logs calls.  Anyone else intrested?  For now I have to do
more clean up... ya know make it pretty. :)

bkw
PS unixODBC is LGPL :)
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Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)

[Tilghman Lesher]

On Tuesday 25 November 2003 03:19, Roy Sigurd Karlsbakk wrote:
> If it was possible to get any support at all from Jeremy (or others),
> I'd be glad to use it. I have sent numerous reports with where it
> failed and what I did to remedy this without getting any response.
> With developers this arrogant, chan_h323 should be removed from the
> asterisk tree, unless the code support is improved.

Please keep your personal attacks off the mailing list.  Jeremy's post
was quite clear about seeking _technical_ issues, not personal issues.
Unless you have specific examples of the shortcomings of that driver,
please do not cry over spilt milk on this mailing list.

-Tilghman

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Re: [Asterisk-Users] cdr_unixodbc

[Brian West]

http://www.bkw.org/~brian/cdr_unixodbc.tar.gz

asterisk root # cd /usr/src/

asterisk src # tar zxfv cdr_unixodbc.tar.gz
cdr_unixodbc/
cdr_unixodbc/cdr_unixodbc.c
cdr_unixodbc/Makefile
cdr_unixodbc/mkdep
cdr_unixodbc/cdr_unixodbc.conf.sample

asterisk src # cd cdr_unixodbc
asterisk cdr_unixodbc # make
./mkdep -fPIC -I../asterisk `ls *.c`
gcc -fPIC -I../asterisk   -c -o cdr_unixodbc.o cdr_unixodbc.c
gcc -shared -Xlinker -x -o cdr_unixodbc.so cdr_unixodbc.o -lodbc -lz
-fPIC -I../asterisk

asterisk cdr_unixodbc # make install
for x in cdr_unixodbc.so; do install -m 755 $x /usr/lib/asterisk/modules ;
done

asterisk cdr_unixodbc #cp cdr_unixodbc.conf.sample to /etc/asterisk/cdr_unixodbc.conf

vi /etc/asterisk/cdr_unixodbc.conf

set the dsn, username, password from your odbc.ini file for the database
you setup fro asterisk cdr.

Also if you find a bug or can do something better in the code please feel
free to do so.  It took all of 3 hours total to write this(most of the
time was finding unixODBC docs.. jesus they say * has bad docs... just try
to find unixODBC docs).  So i'm sure its not without its bugs.

bkw

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RE: [Asterisk-Users] cdr_unixodbc

[Brian West]

I called it that because i'm using the unixODBC libs. I guess I can change
that! :P  I just posted the code and install instructions to the list.

Also if i'm thinking correct this will sidestep the issue with mysql and
gpl since unixODBC is lgpl?

bkw

On Tue, 25 Nov 2003, Vledder, Hans wrote:

> Hi Brian,
>
> Excellent job, but how about calling the application 'cdr_odbc' instead of
> 'cdr_unixodbc', because up to now 'unix' is obvious/trivial when it comes to
> * isn't it? Besides, I think 'cdr_odbc' is more in line with cdr_mysql and
> cdr_csv and what have you ...
>
> Keep it up !
> Hans
>
> -Original Message-
> From: Brian West [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, November 25, 2003 9:28 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] cdr_unixodbc
>
>
> asterisk*CLI> load cdr_unixodbc.so
>  Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR Backend)
>   == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing
> '/etc/asterisk/cdr_unixodbc.conf': Found
> -- cdr_unixodbc: dsn is MySQL-asterisk
> -- cdr_unixodbc: username is root
> -- cdr_unixodbc: password is [secret]
> -- Connected to MySQL-asterisk
>
> it works.. it logs calls.  Anyone else intrested?  For now I have to do
> more clean up... ya know make it pretty. :)
>
> bkw
> PS unixODBC is LGPL :)
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> it to anyone else. If you received it in error please notify us immediately
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Re: [Asterisk-Users] Cisco to use * as a gateway?

[Pavel Litvinenko]

Joseph Finley wrote:

I'm not sure if I am wording this correctly, but I'll try.

I have a Cisco 2621 w/ a couple FXO and FXS ports.  I have a couple cheap
analog phones plugged into the FXS ports.  I am able to get * to ring those
phones when a call comes in, but I cannot get the phones to dial out.  I
guess it's all syntax that I'm doing wrong.  Does someone have a couple
small snip-its to accomplish this?
 

what protocol do you use ? H323, SIP ?

Thanks
Joe
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sip:[EMAIL PROTECTED]


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Re: [Asterisk-Users] cdr_unixodbc

[Tilghman Lesher]

On Tuesday 25 November 2003 07:50, Vledder, Hans wrote:
> Excellent job, but how about calling the application 'cdr_odbc'
> instead of 'cdr_unixodbc', because up to now 'unix' is
> obvious/trivial when it comes to * isn't it? Besides, I think
> 'cdr_odbc' is more in line with cdr_mysql and cdr_csv and what have
> you ...

On the contrary, he's using the unixodbc library, not the iODBC library
(also for Linux), so in case somebody ever creates that cdr driver, it
will greatly lessen the confusion.

-Tilghman

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Re: [Asterisk-Users] * Configuration

[e-smith]

- Original Message - 
From: "Girish Gopinath" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 25, 2003 10:22
Subject: [Asterisk-Users] * Configuration


> Hi,
>
> I am a beginner to Asterisk. Can anybody clear my following doubts
regarding
> the configuration needed?
>
> 1) What is the ideal system configuratin required?(like processer, RAM,
h/d > space etc)
http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware%20recommendations

> 2) How many connections it can handle at a time?
> 3) How many Virtual PBXs it can handle?
> 4) Whether Postgres or Mysql is best suited?
> 5) How many IVR's it can handle simultaneously?
> 6) How many Voicemails can be recorded at a time?
> 7) What type of bandwidth does * require?
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html

>
> Thanx and Regards...
>
> Girish Gopinath


Check the url's.


/Mats

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Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download

[Dan]

Hi,

- Original Message - 
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 25, 2003 3:37 PM
Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step
forward - available for download


>
> Dan
> I seem to be having the same problem as some of the other guys.  With all
> the previous versions I could make outgoing and receive incoming calls;
> however with this latest version even if I have Diax open the call drops
> through to the busy priority in my extensions.conf file.  It's like it's
> showing Diax is already on a call but this is not the case.  If you have
> some suggestions I'll play around with it a bit but something just doesn't
> seem right with it.
> AJ

Together with 0.9.5 wil be a second version of the DLL for IAX(1), till this
issue will be solved in the library.
You will be able to use one of them, as you wish.

Thank you for your understanding,
Dan

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Re: [Asterisk-Users] cdr_unixodbc

[pat munis]

Brian
   Good job!!! Is there any perfomance hit by using unixodbc as oppossed to for 
example using cdr_mysql for mysql?



- Original Message -
From: Brian West <[EMAIL PROTECTED]>
Date: Tue, 25 Nov 2003 07:19:27 -0600 (CST)
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] cdr_unixodbc

> Ok the basic requirement is unixODBC and the MyODBC driver(for MySQL) or
> you can pick what ever you want(http://www.unixodbc.org/drivers.html).
> The table structure is the same as pgsql or mysql ... just duplicate that.
> 
> I would like to verify that I have done this in such a way that the
> database you choose to use won't matter(unixODBC is all about that)
> 
> odbc.ini info:
> 
> [ODBC Data Sources]
> MySQL-asterisk = MySQL ODBC Driver Asterisk DSN
> 
> [MySQL-asterisk]
> Description = MySQL ODBC Driver Asterisk DSN
> Driver  = MySQL
> Socket  = /var/run/mysqld/mysqld.sock
> Server  = localhost
> User= root
> Database= asterisk
> Option  = 3
> #Port   =
> Password= password
> 
> odbcinst.ini
> 
> [MySQL]
> Description = MySQL ODBC MyODBC Driver
> Driver  = /usr/lib/libmyodbc3.so
> 
> ...  I will give you guys time to do that ... and i'm still cleaning up
> the code a bit more.  unixODBC is a bit more forgiving than the MySQL C
> API is.
> 
> 
> bkw
> 
> 
> On Tue, 25 Nov 2003, WipeOut wrote:
> 
> > Pavel Litvinenko wrote:
> >
> > > Brian West wrote:
> > >
> > >> asterisk*CLI> load cdr_unixodbc.so
> > >> Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR
> > >> Backend)
> > >>  == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing
> > >> '/etc/asterisk/cdr_unixodbc.conf': Found
> > >>-- cdr_unixodbc: dsn is MySQL-asterisk
> > >>-- cdr_unixodbc: username is root
> > >>-- cdr_unixodbc: password is [secret]
> > >>-- Connected to MySQL-asterisk
> > >>
> > >> it works.. it logs calls.  Anyone else intrested?  For now I have to do
> > >> more clean up... ya know make it pretty. :)
> > >>
> > >>
> > > Yes,  I do :) where can I get it  ?
> > >
> > What are the software requirements?
> >
> > I have never used unixodbc before.. :)
> >
> > Later..
> >
> > ___
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> >
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RE: [Asterisk-Users] How to use * to simply skim off callerid (UK)?

[Glenn B. Lawler]

Dave,

> I'm wondering how I could go about placing * into an existing office PBX
> system so as to capture callerid for further processing via AGI into an
> intranet app.

If I understand you, all you want to do is use an incoming callerid to trigger
an event in your system. If I got that right, it sounds like attempting to hang
a second PBX system on the incoming phone lines would not only be
overkill, but may result in complications with the incoming calls. We have
developed software in the past that uses a callerid capable modem to
capture callerid data without interfering with the downstream phone system.
That is the approach I would look into; it is not only simpler, but far cheaper
since you only need one modem per incoming line.

HTH,

- Glenn Lawler

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Re: [Asterisk-Users] cdr_unixodbc

[Brian West]

Pat,
What i'm trying to figure out is how to keep the sql statement
globally prepaired then just call SQLExecute but the docs for all this are
hard to come by.  I really can't tell much diffrence in odbc over mysql in
speed but I don't have a bazillion calls going thru at once.  It does add
a layer of abstraction that should allow it to work with just bout any DB
server that has a driver for unixODBC

These include:
MySQL
Postgres
Oracle
DB2
and more...  http://www.unixodbc.org/drivers.html


bkw


On Tue, 25 Nov 2003, pat munis wrote:

> Brian
>Good job!!! Is there any perfomance hit by using unixodbc as oppossed to for 
> example using cdr_mysql for mysql?
>
>
>
> - Original Message -
> From: Brian West <[EMAIL PROTECTED]>
> Date: Tue, 25 Nov 2003 07:19:27 -0600 (CST)
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] cdr_unixodbc
>
> > Ok the basic requirement is unixODBC and the MyODBC driver(for MySQL) or
> > you can pick what ever you want(http://www.unixodbc.org/drivers.html).
> > The table structure is the same as pgsql or mysql ... just duplicate that.
> >
> > I would like to verify that I have done this in such a way that the
> > database you choose to use won't matter(unixODBC is all about that)
> >
> > odbc.ini info:
> >
> > [ODBC Data Sources]
> > MySQL-asterisk = MySQL ODBC Driver Asterisk DSN
> >
> > [MySQL-asterisk]
> > Description = MySQL ODBC Driver Asterisk DSN
> > Driver  = MySQL
> > Socket  = /var/run/mysqld/mysqld.sock
> > Server  = localhost
> > User= root
> > Database= asterisk
> > Option  = 3
> > #Port   =
> > Password= password
> >
> > odbcinst.ini
> >
> > [MySQL]
> > Description = MySQL ODBC MyODBC Driver
> > Driver  = /usr/lib/libmyodbc3.so
> >
> > ...  I will give you guys time to do that ... and i'm still cleaning up
> > the code a bit more.  unixODBC is a bit more forgiving than the MySQL C
> > API is.
> >
> >
> > bkw
> >
> >
> > On Tue, 25 Nov 2003, WipeOut wrote:
> >
> > > Pavel Litvinenko wrote:
> > >
> > > > Brian West wrote:
> > > >
> > > >> asterisk*CLI> load cdr_unixodbc.so
> > > >> Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR
> > > >> Backend)
> > > >>  == Parsing '/etc/asterisk/cdr_unixodbc.conf':   == Parsing
> > > >> '/etc/asterisk/cdr_unixodbc.conf': Found
> > > >>-- cdr_unixodbc: dsn is MySQL-asterisk
> > > >>-- cdr_unixodbc: username is root
> > > >>-- cdr_unixodbc: password is [secret]
> > > >>-- Connected to MySQL-asterisk
> > > >>
> > > >> it works.. it logs calls.  Anyone else intrested?  For now I have to do
> > > >> more clean up... ya know make it pretty. :)
> > > >>
> > > >>
> > > > Yes,  I do :) where can I get it  ?
> > > >
> > > What are the software requirements?
> > >
> > > I have never used unixodbc before.. :)
> > >
> > > Later..
> > >
> > > ___
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> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > ___
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>
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Re: [Asterisk-Users] Re: E100P driver overwrites memory used bye linux-kernel

[Mark Spencer]

> Then i started asterisk, it opens the D-Channel and
> everything is still ok. I left the system in this state
> and it survived one night without problems. But immediately
> after the first call (B-Channel) the system memory is overwritten
> and bad things happen.
> I looked through the wct1xxp.c and it looks like the D-Channel
> (HDLC) is handled between hardware and driver like the B-Channels
> (no HDLC decoding in hardware), right?
> That would mean that it is probably software problem. Is there a
> way to do some test calls without asterisk?

This sounds like mismatched modules, *or* that you have MMX turned on on a
machine that doesn't like MMX or that you're running on a SMP machine
without turning SMP on.

Mark

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Re: [Asterisk-Users] SAY NUMBER in AGI?

[Mark Spencer]

You're forgetting to answer the line first.

Mark

On Fri, 21 Nov 2003, WipeOut wrote:

> I am trying to use the SAY NUMBER command from an AGI script but it does
> not seem to be working..
>
> If I use "EXEC SayNumber 2" and execute the asterisk command from the
> AGI it works and I hear the 2 said on the phone..
>
> If I use "SAY NUMBER 2" I see "-- Playing 'digits/2' (language 'en')" on
> the console but I don't hear the number said on the phone..
>
> I would prefer to use the AGI commands from within the AGI script.. :)
>
> Is this a bug or am I doing something wrong??
>
> Thanks..
>
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Re: [Asterisk-Users] can't get caller id?

[Mark Spencer]

> DTMF is used in some places. Japan uses FSK, but a rather different
> message format. There isn't a whole lot of global standardisation in CLI!

Not only that but I believe they use different frequencies, and utilize a
parity bit as well.

Mark

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Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls

[Andrew Kohlsmith]

> Yep, we use it for international calling.  Works great:
> exten => _9011.,1,Dial(Zap/g0/${EXTEN:1},,t)

How are you achieving that?  If I am on a regular FXS connected phone that 
line would match 90115, thus preventing me from getting the rest of the 
phone number (620508132).

Now if dialing from a SIP or IAX phone, it's no problem since the entire 
number is sent at once.

Regards,
Andrew
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RE: [Asterisk-Users] How to use * to simply skim off callerid (UK)?

[Dave Wilson]

>> I'm wondering how I could go about placing * into an existing office PBX
>> system so as to capture callerid for further processing via AGI into an
>> intranet app.
>
>If I understand you, all you want to do is use an incoming callerid to
trigger
>an event in your system. If I got that right, it sounds like attempting to
hang
>a second PBX system on the incoming phone lines would not only be
>overkill, but may result in complications with the incoming calls. We have
>developed software in the past that uses a callerid capable modem to
>capture callerid data without interfering with the downstream phone system.
>That is the approach I would look into; it is not only simpler, but far
cheaper
>since you only need one modem per incoming line.

Glenn,

Has your previously mentioned software been compatible with UK (BT)
callerid? I realise Asterisk is overkill for what I've described, however it
was the only real solution I was able to find to allow us to develop the
interface we were looking for.  Though I've said we are only concerned with
the callerid at the moment, this is purely for initial installs. Once we
have our system installed, then we start a program of adding modules, some
of which are tightly integrated with Asterisk functionality (intelligent
call routing based on workflow management rules). We simply want to keep
initial install costs to a minimum for our clients as its difficult enough
to force them to buy in to a complete dealer management solution, without
also having to persuade them to dump their existing telephony systems too.

Thanks,
Dave


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Re: [Asterisk-Users] * Configuration

[Joe Kellman]

befor answering these questions, ask yourself the
following questions:

what is it that you want to achieve with your
deployment?  

How many users are you planning to service with
Asterisk?

Are you trying to integrate with an existing PBX?

If this installation is a stand-alone, how do you wish
to connect to PSTN?

there are other questions the come to mind but don't
have time to write them down...but i think this should
get you started.  also take a look at the asterisk
reference at http://www.voip-info.org...jak


--- Girish Gopinath <[EMAIL PROTECTED]>
wrote:
> Hi,
> 
> I am a beginner to Asterisk. Can anybody clear my
> following doubts regarding
> the configuration needed?
> 
> 1) What is the ideal system configuratin
> required?(like processer, RAM, h/d 
> space etc)
> 2) How many connections it can handle at a time?
> 3) How many Virtual PBXs it can handle?
> 4) Whether Postgres or Mysql is best suited?
> 5) How many IVR's it can handle simultaneously?
> 6) How many Voicemails can be recorded at a time?
> 7) What type of bandwidth does * require?
> 
> Thanx and Regards...
> 
> Girish Gopinath
> 
>
_
> The Great MSN Sale. Get shopping discounts.
> http://www.msn.co.in/Shopping 
> Win exciting prizes!
> 
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[Asterisk-Users] music on hold

[zoa]

Sometimes when people hang up, or the call gets interrupted for some 
reason, music on hold starts playing. (i use app_dial without extra 
parameters and no moh is set in extensions.conf)

Any suggestions on how i could get rid of this 'feature' ?

greetz,

zoa.

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[Asterisk-Users] Prompt recording

[Jerimiah Cole]

Does anybody have useful tips on creating good quality recordings for 
use with prompts in asterisk?  I'm interested in hearing input on 
hardware (mics, dats, sound cards, etc) and software (recording 
software, dsp) as well as recording techniques.

Jerimiah
Tularosa Communications
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Re: [Asterisk-Users] Echo cancellation

[Peter Zeltins]

Hi,

I'm interested. I'm running chan_capi 0.3.0 with Fritz PCI ISDN card. Using
DIAX as softphone and dialing out to PSTN generally results in good sound
quality at softphone end (no echo), but PSTN end experiences quite a bit of
echo. I have enabled echosquelch in capi.conf, but it does not seem to help


thx,
Peter

> > What is the status on echo cancellation in Asterisk/CAPI?
>
> I tried to straighten this out; I think it works, but I'm not sure.  If
> it does work, I think it might find its way in future chan_capi
> releases; if you want to try it out at this early stage, contact me
> off-list!
>
> --
> Emanuele
>
>

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RE: [Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download

[Florian Overkamp]

Hi, 

> -Original Message-
> check both directions
> when you do a show channel ... does it show gsm in both ways ?

Yes:

vectra*CLI> show channel IAX2[florian]/14
 -- General --
   Name: IAX2[florian]/14
   Type: IAX2
   UniqueID: 1069774221.220
  Caller ID: 651154495
DNID Digits: (N/A)
  State: Up (6)
  Rings: 0
   NativeFormat: 8
WriteFormat: 8
 ReadFormat: 8
1st File Descriptor: -1
  Frames in: 786
 Frames out: 1778
 Time to Hangup: 0
 --   PBX   --
Context: onnet-basic
  Extension: s
   Priority: 1
 Call Group: 0
   Pickup Group: 0
Application: Bridged Call
   Data: Zap/62-1
  Stack: -1
Blocking in: ast_waitfor_nandfds

(and the same for Zap/62-1)

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RE: [Asterisk-Users] Picking a channel (FXO port or SIP) for outb ound calls

[Tony Kava]

> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, 25 November, 2003 08:56
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for
outbound calls
> 
> 
> > Yep, we use it for international calling.  Works great:
> > exten => _9011.,1,Dial(Zap/g0/${EXTEN:1},,t)
> 
> How are you achieving that?  If I am on a regular FXS connected phone that

> line would match 90115, thus preventing me from getting the rest of the 
> phone number (620508132).
> 
> Now if dialing from a SIP or IAX phone, it's no problem since the entire 
> number is sent at once.
> 
> Regards,
> Andrew

This works fine for me on my analog phones using a Digium FXS.  I believe
that '.' will match any remaining part of the string.  I don't believe it
matches like a regular expression would (any single character).

--
Tony Kava
Network Administrator
Pottawattamie County, Iowa
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RE: [Asterisk-Users] Picking a channel (FXO port or SIP) for outb ound calls

[Tony Kava]

> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, 25 November, 2003 08:56
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for
outbound calls
> 
> 
> > Yep, we use it for international calling.  Works great: exten => 
> > _9011.,1,Dial(Zap/g0/${EXTEN:1},,t)
> 
> How are you achieving that?  If I am on a regular FXS connected phone 
> that
> line would match 90115, thus preventing me from getting the rest of the 
> phone number (620508132).
> 
> Now if dialing from a SIP or IAX phone, it's no problem since the 
> entire
> number is sent at once.
> 
> Regards,
> Andrew

I double-checked, and I do have a timeout on that.  My previous message may
have been mistaken.

--
Tony Kava
Network Administrator
Pottawattamie County, Iowa
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Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls

[marrandy]

On Tuesday 25 November 2003 09:56 am, Andrew Kohlsmith wrote:
> > Yep, we use it for international calling.  Works great:
> > exten => _9011.,1,Dial(Zap/g0/${EXTEN:1},,t)
> 
> How are you achieving that?  If I am on a regular FXS connected phone that 
> line would match 90115, thus preventing me from getting the rest of the 
> phone number (620508132).


Easy...   _9 get's an outside line.  011 is the USA international access 
number.  . (period) is match anything else.

Please show the line (cut and paste) you are presently using.

You are not helping anyone fix your problem by omitting it.

Regards...Martin
-- 
Computer Science is merely the post-Turing decline in formal systems theory.

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Re: [Asterisk-Users] Can you monitor a call via the asterisk speaker system and do a call pickup if you wish

[marrandy]

On Friday 21 November 2003 12:36 pm, marrandy wrote:
> Reason.
> 
> I have a fax/ans phone with handset, that lets you monitor the caller, so if 
> you wish, you can pickup the call.
> 
> The asterisk is undergoing testing, it will then be online tested at the 
house 
> so I can get more familiar in setting components up, e.g. sip phones, 
> voicemail, transfers etc.
> 
> But, I really need the monitoring of the Voicemall being left, with the 
> ability to pick up the call, otherwise, someone is going to be unhappy.
> 
> If this functionality isn't available.  Can it be added as a request to the 
> developers.
> 
> Regards...martin


Well no reply yet, so I've been further searching on the assumption that this 
is so obvious, I should have found the answer myself.

Despite some interesting related threads, I'm still none the wiser.

Surely, there is an option to monitor the Voicemail either at the the console 
via the speakers and pick it up.

It would also be handy to monitor at the extension (speakerphone) and pickup 
by punching in a code, or just picking up the extension.. For example, you 
may be busy doing something else, decide to let VM handle a call, but on 
hearing who the message is from, or, what the subject is about, decide you 
want to deal with it immediately.
Often, if it's out of area ie. no callerID.

Any feedback appreciated.

Regards...Martin
-- 
aphorism, n.:
A concise, clever statement.
afterism, n.:
A concise, clever statement you don't think of until too late.
-- James Alexander Thom

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RE: [Asterisk-Users] zt_rec: Unknown error 500

[Scott Stingel]

Hi Michiel-

This may be related to a PRI frame buffer overflow problem that I get in
high-volume IVR applications.  I get a lot of these errors mixed in with
frame errors.   In my case its load related.  Mark and Martin at Digium have
said they'll be looking into improving the buffering mechanism.
 

-Scott

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]    
URL:www.evtmedia.com    



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michiel Betel
> Sent: Tuesday, November 25, 2003 1:43 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] zt_rec: Unknown error 500
> 
> 
> I have a number of Zap/ extensions defined in a queue with ringall 
> strategy. When this queue is called sometimes Asterisk seems to think 
> that one of these channels is busy, while it is NOT. The following is 
> shown on the console:
>  --Called 44
> -- Called 36
> -- Called 41
> -- Called 35
> -- Called 38
> -- Zap/44-1 is ringing
> -- Zap/36-1 is ringing
> -- Zap/41-1 is ringing
> -- Zap/35-1 is ringing
> -- Zap/38-2 is ringing
> -- Zap/44-1 is ringing
> -- Zap/36-1 is ringing
> -- Hungup 'Zap/35-1'
> -- Zap/41-1 is ringing
> -- Zap/44-1 is ringing
> -- Zap/36-1 is ringing
> While a
> WARNING[165916]: File chan_zap.c, Line 3331 (zt_read): 
> zt_rec: Unknown 
> error 500
> is generated in /var/log/asterisk/messages
> Any ideas on how to fix this?? Thanks!
> 
> Michiel
> 
> 
> 
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Re: [Asterisk-Users] cdr_unixodbc

[Brian West]

Just an FYI I have cdr_unixodbc doing inserts using Text file driver
now

bkw
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[Asterisk-Users] (no subject)

[Antonio Sanz]

Hi,

First at alll, I beg your pardon because maybe I explained bad my questions 
(because my low level english)

I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2

compiled and installed.

I have modules alsa 0.9.8 compiled and installed

My PC has an audio card ac97 chipset intel i810 in its motherboard.

I want to use asterisk in this way:

|PC linux |  --ethernet--  |asterisk|  --internet--  |gatekeeper|  -- |PSTN|
 |
|PC win| -ethernet---|
A PC's network connected to * via ethernet with gnophone clients or 
iaxcomm.ASterisk is connected to a gatekeeper and the gatekeeper sends the 
calls to the PSTN.

In order to do that, maybe I don't need soundcard in *. Coul you please 
confirm it to me?

In the other hand. in astersk configuration examples there are references 
to calls that finish in the PBX and can be used as voice mail or to give 
anouncements or simply to make an operator hang-off. I understand that a 
soundcard is needed for some of these options

When I ran asterisk loading alsa module. * gave me an error, and finished. 
Now, I ran it with module oss and it starts, but it gives me this warning

[chan_oss.so] => (OSS Console Channel Driver)
== Console is full duplex
== Registered channel type 'Console' (OSS Console Channel Driver)
== Parsing '/etc/asterisk/oss.conf': Found
[res_adsi.so]WARNING[3076]: File chan_oss.c, Line 232 (sound_thread): Read
error on sound device: Resource temporarily unavailable
I don't know how to to do that soundcard i8x0 works with OSS, I only know 
to make it work with ALSA, but for the tasks I want to do maybe I don't 
need soundcard, for that reason I would like to comment another problem I 
have in the configuration I am testing

In the configuration file 0h323 I have defined a gatekeeper, and when * 
starts, it is well registered in that gatekeeper.

When I make a call since gnophone (Linux) or iaxcomm (win98) to a PSTN 
telephone number, the call goes well, the PSTN telephone rings, but when I 
hang-off nothing is heard in any address.

The * call logis:

CLI> -- Accepting AUTHENTICATED call from 10.16.96.149, requested format
= 2, actual format = 2
-- Executing Dial("[EMAIL PROTECTED]:5036]/1",
"OH323/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- H323:25026 answered [EMAIL PROTECTED]:5036]/1
-- Hungup 'H323:25026'
== Spawn extension (default, 913014647, 1) exited non-zero on
'[EMAIL PROTECTED]:5036]/1'
-- Hungup '[EMAIL PROTECTED]:5036]/1'
Un the file extensions.conf I have an input for all te calls to go through 
the gatekeeper:

exten => _91XXX,1,Dial,OH323/[EMAIL PROTECTED]

I don't know if it is mandatory to define here the gatekeeper address 
(although I have already done in the oh323.conf), but in this same 
distribution list I read this configuration example (with this one, the 
call works, but nothing is heard):

When the call is done, I see packets between gnophone and asterisk 
bidirectionally: 0.16.96.148 (asterisk)   10.16.96.149(gnophone)

namor:/home/sanz# tcpdump -n -i eth0 udp and host 10.16.96.148
tcpdump: listening on eth0
Estabishment phase:

17:36:55.650506 10.16.96.149.5036 > 10.16.96.148.5036:  udp 104 (DF) [tos0x10]
17:36:55.654766 10.16.96.149.5036 > 10.16.96.148.5036:  udp 12 (DF) [tos 0x10]
17:36:55.674871 10.16.96.148.5036 > 10.16.96.149.5036:  udp 12 (DF) [tos 0x10]
17:36:55.675328 10.16.96.148.5036 > 10.16.96.149.5036:  udp 69 (DF) [tos 0x10]
17:36:55.675504 10.16.96.148.5036 > 10.16.96.149.5036:  udp 12 (DF) [tos 0x10]
17:36:55.676650 10.16.96.149.5036 > 10.16.96.148.5036:  udp 12 (DF) [tos 0x10]
17:36:55.690271 10.16.96.148.5036 > 10.16.96.149.5036:  udp 12 (DF) [tos 0x10]
17:36:55.691258 10.16.96.149.5036 > 10.16.96.148.5036:  udp 12 (DF) [tos 0x10]
17:37:00.314595 10.16.96.149.5036 > 10.16.96.148.5036:  udp 55 (DF) [tos 0x10]
17:37:00.314758 10.16.96.148.5036 > 10.16.96.149.5036:  udp 12 (DF) [tos 0x10]
17:37:00.315792 10.16.96.148.5036 > 10.16.96.149.5036:  udp 23 (DF) [tos 0x10]
17:37:00.316550 10.16.96.149.5036 > 10.16.96.148.5036:  udp 12 (DF) [tos 0x10]
17:37:00.881438 10.16.96.148.5036 > 10.16.96.149.5036:  udp 12 (DF) [tos 0x10]
17:37:00.882394 10.16.96.149.5036 > 10.16.96.148.5036:  udp 12 (DF) [tos 0x10]
17:37:00.887593 10.16.96.149.5036 > 10.16.96.148.5036:  udp 45 (DF) [tos 0x10]
17:37:00.887812 10.16.96.148.5036 > 10.16.96.149.5036:  udp 12 (DF) [tos 0x10]
17:37:00.890902 10.16.96.148.5036 > 10.16.96.149.5036:  udp 45 (DF) [tos 0x10]
17:37:00.891565 10.16.96.149.5036 > 10.16.96.148.5036:  udp 12 (DF) [tos 0x10]
voice

17:37:00.904894 10.16.96.149.5036 > 10.16.96.148.5036:  udp 37 (DF) [tos 0x10]
17:37:00.910874 10.16.96.148.5036 > 10.16.96.149.5036:  udp 37 (DF) [tos 0x10]
17:37:00.922539 10.16.96.149.5036 > 10.16.96.148.5036:  udp 37 (DF) [tos 0x10]
17:37:00.931519 10.16.96.148.5036 > 10.16.96.149.5036:  udp 37 (DF) [tos 0x10]
17:37:00.940179 10.16.96.149.5036 > 10.16.96.148.5036:  udp 37 (DF) [tos 0x10]
17:37:00.951466

[Asterisk-Users] PCI 3.3 V

[Cristian Vasiliu]

Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find 
any motherboard with PCI 3.3 . Any sugestions!?

Cristian VASILIU
AccessNET International S.A.
Software Programmer
mail to :<[EMAIL PROTECTED]>
www:
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Re: [Asterisk-Users] Ring requested on channel 1 already in use...

[Martin Pycko]

Do you have up to date libpri and asterisk ?

Also it'd be good if you could send "pri debug span 1" (or 2) trace.

regards
Martin

On Tue, 25 Nov 2003, Alastair Maw wrote:

> I'm running an E400P. Every now and then Asterisk stops receiving
> incoming calls.
>
>
>
> This turns up in the messages log:
>
> Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793
> (pri_dchannel): Ring requested on channel 1 already in use on span 1.
> Hanging up owner.
>
> Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793
> (pri_dchannel): Ring requested on channel 1 already in use on span 2.
> Hanging up owner.
>
> Nov 25 10:49:25 WARNING[98311]: File chan_zap.c, Line 5793
> (pri_dchannel): Ring requested on channel 1 already in use on span 3.
> Hanging up owner.
>
> Nov 25 10:49:25 WARNING[114696]: File chan_zap.c, Line 5793
> (pri_dchannel): Ring requested on channel 1 already in use on span 4.
> Hanging up owner.
>
>
>
> A little while back I also had this in my logs:
>
> Nov 15 17:25:21 WARNING[114696]: File chan_zap.c, Line 5790
> (pri_dchannel): Duplicate setup requested on channel 11 already in use
> on span 4
>
> Nov 15 17:25:21 WARNING[65541]: File chan_zap.c, Line 5790
> (pri_dchannel): Duplicate setup requested on channel 4 already in use on
> span 1
>
> Nov 15 17:25:21 WARNING[114696]: File chan_zap.c, Line 5790
> (pri_dchannel): Duplicate setup requested on channel 12 already in use
> on span 4
>
> Nov 15 17:25:21 WARNING[65541]: File chan_zap.c, Line 5790
> (pri_dchannel): Duplicate setup requested on channel 5 already in use on
> span 1
>
> Nov 15 17:25:22 WARNING[65541]: File chan_zap.c, Line 5790
> (pri_dchannel): Duplicate setup requested on channel 3 already in use on
> span 1
>
> Nov 15 17:25:22 WARNING[65541]: File chan_zap.c, Line 5790
> (pri_dchannel): Duplicate setup requested on channel 2 already in use on
> span 1
>
> Nov 15 17:25:24 WARNING[114696]: File chan_zap.c, Line 5790
> (pri_dchannel): Duplicate setup requested on channel 13 already in use
> on span 4
>
>
> FWIW, my libpri/zaptel/asterisk installs are all about two months old.
> Might whatever causes this have been fixed by now? (I don't want to
> upgrade otherwise as this problem is quite intermittent).
>
> Anyone have any ideas?
>
> Alastair
>
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Re: [Asterisk-Users] Strange code in rtp.c / disconnect - maybe reinvite problems

[Martin Pycko]

OK, that was obviously a 'typo' ... It's fixed.

Martin

On Tue, 25 Nov 2003, Detlef Wengorz wrote:

> Daniel Chabrol wrote:
> >
> > Hi List!
> >
> > I get "WARNING[14351]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 =
> > 524300 is not codec1 = 524300, can't do reinvite" at my asterisk console.
> >
> > The code there looks realy strange:
> >
> > codec0 = pr0->get_codec(c0);
> > codec1 = pr1->get_codec(c1);
> > ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, can't do
> > reinvite\n",codec0,codec1);
> > /* Hey, we can't do reinvite if both parties speak diffrent codecs */
> > if (codec0 != codec1)
> > return -2;
> >
> > I think the message should only occur *after* checking equality:
> >
> > if (codec0 != codec1) {
> > ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, can't do
> > reinvite\n",codec0,codec1);
> > return -2;
> > }
> >
> > I hoped this "can't do reinvite" would explain my disconnects from the
> > nikotel.com sip server after 60 seconds. But this little bug seems only
> > to be display-specific and not affect funtion. But maybe i oversight
>
> That's correct :-(
> but change the code like this
>
> if (codec0 != codec1) {
>  ast_log(LOG_WARNING,
>  "codec0 = %d is not codec1 = %d, can't do
> reinvite\n",codec0,codec1);
>
>  ast_mutex_unlock(&c0->lock); // unlock before return
>  ast_mutex_unlock(&c1->lock); // unlock before return
>  return -2;
> }
>
> and try again.
> maybe it helps.
>
>
>
>
> > something which still disables the reinvite even if i use
> > canreinvite=yes in my sip.conf:
> >
> > [nikotel]
> > type=friend
> > username=USERID
> > fromuser=USERID
> > secret=PASSWORD
> > host=calamar0.nikotel.com
> > canreinvite=yes
> > context=internal
> > ; no nat entry because im not using nat!
> >
> > Is there someone which is able to use Nikotel.com with the current
> > CVS-Version (in my case CVS-11/24/03-19:24:22). BTW: 0.5.0 don't work
> > too in my case (at least not longer than 60 seconds). Pulver.com calls
> > and so on are no problem. Any suggestions?
> >
> > Best regards,
> > Daniel
> >
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Best regards
> Detlef Wengorz <[EMAIL PROTECTED]>
> Detlef Wengorz <[EMAIL PROTECTED]>
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Re: [Asterisk-Users] zt_rec: Unknown error 500

[Michiel Betel]

My Zap channels having the problems are on a T1 connected to a CAC 
channelbank, But it looks like the zt_rec in chan_zap error uses the 
lowlevel zaptel ioctl's which are the same for T1 & PRI...

Scott Stingel wrote:

Hi Michiel-

This may be related to a PRI frame buffer overflow problem that I get in
high-volume IVR applications.  I get a lot of these errors mixed in with
frame errors.   In my case its load related.  Mark and Martin at Digium have
said they'll be looking into improving the buffering mechanism.
 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Michiel Betel
Sent: Tuesday, November 25, 2003 1:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] zt_rec: Unknown error 500

I have a number of Zap/ extensions defined in a queue with ringall 
strategy. When this queue is called sometimes Asterisk seems to think 
that one of these channels is busy, while it is NOT. The following is 
shown on the console:
--Called 44
   -- Called 36
   -- Called 41
   -- Called 35
   -- Called 38
   -- Zap/44-1 is ringing
   -- Zap/36-1 is ringing
   -- Zap/41-1 is ringing
   -- Zap/35-1 is ringing
   -- Zap/38-2 is ringing
   -- Zap/44-1 is ringing
   -- Zap/36-1 is ringing
   -- Hungup 'Zap/35-1'
   -- Zap/41-1 is ringing
   -- Zap/44-1 is ringing
   -- Zap/36-1 is ringing
While a
WARNING[165916]: File chan_zap.c, Line 3331 (zt_read): 
zt_rec: Unknown 
error 500
is generated in /var/log/asterisk/messages
Any ideas on how to fix this?? Thanks!

Michiel

   




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RE: [Asterisk-Users] Sip phones!

[Sérgio Bernardo]

Hi!

I've contacted Grandstream directly via email and received a reply in
one day with prices and an order form to fill... Nice customer service!

--
Sérgio



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Ariel Batista
> Sent: segunda-feira, 24 de Novembro de 2003 19:29
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Sip phones!
> 
> 
> I am trying to get the following phones for testing.  Is 
> there a distributor in the US that is able to sell me these 
> Sip phone and ATA adapters?  I can not afford the Cisco 
> phones there too hard to configure and too expensive!
> 
> 1 - Sipura SPA-2000
> 2 - Grandstream Sip phone BT-102
> 1 - Grandstream HT-286
> 1 - Snom 105 Sip phone.
>  
> I have called and emailed chagres but they have not reply.  
> Nor returned my calls!  I need to test and then deploy these 
> phones and system by the middle of December. 
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Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls

[Tilghman Lesher]

On Tuesday 25 November 2003 08:56, Andrew Kohlsmith wrote:
> > Yep, we use it for international calling.  Works great:
> > exten => _9011.,1,Dial(Zap/g0/${EXTEN:1},,t)
>
> How are you achieving that?  If I am on a regular FXS connected
> phone that line would match 90115, thus preventing me from getting
> the rest of the phone number (620508132).

I'm not achieving anything.  It just works.  If it matters, that line
is directly in the context for the phone as defined in zapata.conf
(context=internal).

-Tilghman

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Re: [Asterisk-Users] PCI 3.3 V

[John Bigelow]

Here are some links.

http://www.supermicro.com/PRODUCT/MotherBoards/GC_SL/X5SSE-G.htm
http://www.supermicro.com/PRODUCT/MotherBoards/GC_SL/X5SS8.htm
http://usa.asus.com/products/server/srv-mb/nrl-ls533/overview.htm
http://www.tyan.com/products/html/trinitygcsl.html

-John

On Tue, Nov 25, 2003 at 06:58:47PM +0200, Cristian Vasiliu wrote:
> Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find 
> any motherboard with PCI 3.3 . Any sugestions!?
> 
> Cristian VASILIU
> AccessNET International S.A.
> Software Programmer
> 
> mail to :<[EMAIL PROTECTED]>
> www:
> 
> 
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Re: [Asterisk-Users] PCI 3.3 V

[Alastair Maw]

On 25/11/03 16:58, Cristian Vasiliu wrote:
Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find 
any motherboard with PCI 3.3 . Any sugestions!?
Wait for the TE405P to appear, which is a 5V version of the TE410P. It 
should be shipping in the next week or two.

Alastair

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Re: [Asterisk-Users] Prompt recording

[Steven Critchfield]

On Tue, 2003-11-25 at 09:24, Jerimiah Cole wrote:
> Does anybody have useful tips on creating good quality recordings for 
> use with prompts in asterisk?  I'm interested in hearing input on 
> hardware (mics, dats, sound cards, etc) and software (recording 
> software, dsp) as well as recording techniques.

Anything that down mixes to 8khz 8bit is fine. 8khz 8bit is the best you
are going to get on a phone line anyways, so shoot for just a tad above,
and accept the down mixing.

BTW, where you to lazy to ask google?
http://www.google.com/search?hl=en&ie=UTF-8&oe=UTF-8&q=sound+quality+recording+site%3Alists.digium.com&btnG=Google+Search
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] PCI 3.3 V

[Tom Walsh]

::Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
::any motherboard with PCI 3.3 . Any sugestions!?
::

This really is a problem with the state of flux the PCI bus is currently in
and the comprimises a vendor must make in order to best meet what is
available in the market (present as well as future).

Dell offers a tower that I know for certain has 3.3V PCI bus. PowerEdge
600SC.

(hope this doesn't wrap)

http://www1.us.dell.com/content/products/productdetails.aspx/pedge_600sc?c=u
s&cs=555&l=en&s=biz

Rackmount... I don't know about... So far... this is one of the only PCs I
have come across that has the 3.3V PCI bus.

Tom Walsh
Network Administrator
http://www.ala.net/

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Re: [Asterisk-Users] PCI 3.3 V

[Sean P. Robertson]

FYI: According to Digium, we should have the new 5v Quad T1/E1/PRI (Part#
TE405P) in stock sometime next week.

Sean
- Original Message - 
From: "Cristian Vasiliu" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 25, 2003 11:58 AM
Subject: [Asterisk-Users] PCI 3.3 V


> Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
> any motherboard with PCI 3.3 . Any sugestions!?
>
> Cristian VASILIU
> AccessNET International S.A.
> Software Programmer
>
> mail to :<[EMAIL PROTECTED]>
> www:
>
>
> ___
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>


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Re: [Asterisk-Users] MGCP RFC (2705) vs. PacketCable MGCP spec

[Mark Spencer]

I think we have to figure out what the difference is.  It'll take going
through the "mgcp debug" output to see what is going on.

Mark

On Mon, 24 Nov 2003, ProvoCityPower wrote:

> We are working on a new implementation of asterisk. We are using a fiber-served 
> WorldWide Packet switch at the home that incorporates a VOIP T2 switch that feeds 2 
> POTS connections. We are told that the T2 is programmed with code that follows the 
> PacketCable spec. This version has a problem with the 'congestion' message which is 
> based on the MGCP RFC (2705) spec. This causes the T2 to become somewhat corrupted 
> and to play a fluctuating buzzing sound (congestion tone?) after a number is 
> pressed. Dial tone is present. A number may be called, but the tone will stay on. It 
> takes a hard reset of the T2 to restore normal service.
>
> Is there any global way to adhere to the PacketCable spec? Do we need to change the 
> extensions.conf file to use something other than congestion? Are there other 
> conflicts that will bite us?
>
> Any input is truly appreciated.
>
> Thanks,
> Jeff

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RE: [Asterisk-Users] PCI 3.3 V

[Scott Stingel]

Compatible motherboards supporting 3.3v PCI are a bit hard to find outside
the US - Tyan makes one (model S2723), and one or two others (Intel?)

But I understand there is a new 5 volt version of the T1/E1 card soon to be
released as well.

-Scott

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]    
URL:www.evtmedia.com    



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Cristian Vasiliu
> Sent: Tuesday, November 25, 2003 4:59 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] PCI 3.3 V
> 
> 
> Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find 
> any motherboard with PCI 3.3 . Any sugestions!?
> 
> Cristian VASILIU
> AccessNET International S.A.
> Software Programmer
> 
> mail to :<[EMAIL PROTECTED]>
> www:
> 
> 
> ___
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> 
> 

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Re: [Asterisk-Users] Can you monitor a call via the asterisk speaker system and do a call pickup if you wish

[Andrew Thompson]

- Original Message - 
From: "marrandy" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 25, 2003 10:47 AM
Subject: Re: [Asterisk-Users] Can you monitor a call via the asterisk speaker system 
and do a call pickup if you wish


> On Friday 21 November 2003 12:36 pm, marrandy wrote:
> > Reason.
> > 
> > I have a fax/ans phone with handset, that lets you monitor the caller, so if 
> > you wish, you can pickup the call.
> > 
> > The asterisk is undergoing testing, it will then be online tested at the 
> house 
> > so I can get more familiar in setting components up, e.g. sip phones, 
> > voicemail, transfers etc.
> > 
> > But, I really need the monitoring of the Voicemall being left, with the 
> > ability to pick up the call, otherwise, someone is going to be unhappy.
> > 
> > If this functionality isn't available.  Can it be added as a request to the 
> > developers.
> > 
> > Regards...martin
> 
> 
> Well no reply yet, so I've been further searching on the assumption that this 
> is so obvious, I should have found the answer myself.
> 
> Despite some interesting related threads, I'm still none the wiser.
> 
> Surely, there is an option to monitor the Voicemail either at the the console 
> via the speakers and pick it up.
> 
> It would also be handy to monitor at the extension (speakerphone) and pickup 
> by punching in a code, or just picking up the extension.. For example, you 
> may be busy doing something else, decide to let VM handle a call, but on 
> hearing who the message is from, or, what the subject is about, decide you 
> want to deal with it immediately.
> Often, if it's out of area ie. no callerID.
> 
> Any feedback appreciated.

Try searching for zapbarge to pick up a call. I think there's a sibling to zapbarge 
that lets you just monitor, but I don't recall the name. hit the archives good, it's 
in there...

> Regards...Martin


-
Andrew [EMAIL 
PROTECTED])fjåŠËbú?jË^®+$ºÇ«

Re: [Asterisk-Users] Prompt recording

[Steve Underwood]

Steven Critchfield wrote:

On Tue, 2003-11-25 at 09:24, Jerimiah Cole wrote:
 

Does anybody have useful tips on creating good quality recordings for 
use with prompts in asterisk?  I'm interested in hearing input on 
hardware (mics, dats, sound cards, etc) and software (recording 
software, dsp) as well as recording techniques.
   

Anything that down mixes to 8khz 8bit is fine. 8khz 8bit is the best you
are going to get on a phone line anyways, so shoot for just a tad above,
and accept the down mixing.
BTW, where you to lazy to ask google?
http://www.google.com/search?hl=en&ie=UTF-8&oe=UTF-8&q=sound+quality+recording+site%3Alists.digium.com&btnG=Google+Search
 

Use 8kHz 16 bits, not 8 bits. The phone line is 12 to 13 bits compressed 
down to 8 in a pseudo-logartihmic way. If you start with 8 bit linear 
data it will sounds considerably worse than 8 bit data, unless the 
volume is very uniform.

When recording find a really quite place. Background noise is usually 
the biggest hassle when recording prompts. Other than that, use a 
reasonable mic; fix it down someplace (don't hand hold it); and get your 
"voice model" to sit comfortably, so they sound nice and relaxed. Choose 
a friend or colleague - someone easily accessible for more recordings 
when you realise you have forgotten some. :-)

Regards,
Steve
Regards,
Steve
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Re: [Asterisk-Users] Re: E100P driver overwrites memory used bye linux-kernel

[Alberto Bertogli]

On Tue, Nov 25, 2003 at 08:51:03AM -0600, Mark Spencer wrote:
> > Then i started asterisk, it opens the D-Channel and
> > everything is still ok. I left the system in this state
> > and it survived one night without problems. But immediately
> > after the first call (B-Channel) the system memory is overwritten
> > and bad things happen.
> > I looked through the wct1xxp.c and it looks like the D-Channel
> > (HDLC) is handled between hardware and driver like the B-Channels
> > (no HDLC decoding in hardware), right?
> > That would mean that it is probably software problem. Is there a
> > way to do some test calls without asterisk?
> 
> This sounds like mismatched modules, *or* that you have MMX turned on on a
> machine that doesn't like MMX or that you're running on a SMP machine
> without turning SMP on.

So running a non-SMP kernel on a SMP machine is fine, but wct1xxp can't
run in an UP kernel on a SMP machine?

This is a very important thing to know because one would assume that as
the kernel is fine, loading the module should be OK too.

Does this restricion apply to any other zapata modules?

How about HT?


Thanks,
Alberto


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[Asterisk-Users] Outgoing-call and enter user in Conference - repost

[Areski]

Hi all,


Just wondering if someone have already done something like that :


SIP Client_A ---> 1)call --->  ASTERISK  ---> 2)outgoingcall-PSTN-->Client_B
 |
 |
   3) Enter conference   | 
 MeetMe <'
   with user A


Make 2 user in conference (point 1 and 2), it's definitely easy, but call an other user
and put the both in conference,I still don't have any idea how to do it!


Thanks in advance for your help,
Areski

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Re: [Asterisk-Users] Prompt recording

[Sri]

I used windows sound recorder to record it (with noise less background)
used sox to convert to gsm. It turned out pretty good.
Jerimiah Cole wrote:

Does anybody have useful tips on creating good quality recordings for 
use with prompts in asterisk?  I'm interested in hearing input on 
hardware (mics, dats, sound cards, etc) and software (recording 
software, dsp) as well as recording techniques.

Jerimiah
Tularosa Communications
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Re: [Asterisk-Users] PCI 3.3 V

[Steve Underwood]

Cristian Vasiliu wrote:

Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find 
any motherboard with PCI 3.3 . Any sugestions!? 
You have four options:

A 32 bit slot in a Dell 600SC
Almost any 64bit PCI slot (except for a small number of 33MHz only 64 
bit slots)
Wait for the soon to arive updated card that can run at 5V
Abandon the telecoms buisiness.

.Its your choice, but option 3 might look good :-) Digium 
miscalculated the market's movement to 3.3V, but they are correcting 
that now with a revisied card.

Regards,
Steve
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Re: [Asterisk-Users] Cisco to use * as a gateway?

[Andrew Gillham]

Pavel Litvinenko wrote:

Joseph Finley wrote:

I'm not sure if I am wording this correctly, but I'll try.

I have a Cisco 2621 w/ a couple FXO and FXS ports.  I have a couple 
cheap
analog phones plugged into the FXS ports.  I am able to get * to ring 
those
phones when a call comes in, but I cannot get the phones to dial out.  I
guess it's all syntax that I'm doing wrong.  Does someone have a couple
small snip-its to accomplish this?
 


This is what a buddy of mine uses to call my pbx extensions.
!
voice service voip
h323
sip
 bind all source-interface FastEthernet0/0   <<--- the public IP interface
!
! The Cisco 7960s only do these two codecs. (also g711alaw)
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
! An analog port on the 3725 router.
voice-port 1/0/0
description POTS Test Phone
!
! The local number for the analog port.
dial-peer voice 100 pots
application session
destination-pattern 6110
port 1/0/0
!
! forward anything 6XXX to my pbx at 1.2.3.4
dial-peer voice 111 voip
preference 1
destination-pattern 6...
voice-class codec 1
voice-class h323 1
session protocol sipv2
session target ipv4:1.2.3.4
ip qos dscp cs5 media
no vad
!
! I believe this just tells where the server is, it doesn't REGISTER.
sip-ua
sip-server ipv4:1.2.3.4
!
Newer Cisco IOS is supposed to be able to register via SIP, but the version
my buddy is running doesn't currently support it.
But he is able to dial my pbx easily, and I can setup the sip.conf with 
a default
ip for his router, etc.

-Andrew

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Re: [Asterisk-Users] PCI 3.3 V

[Amaury Jacquot]

Tom Walsh wrote:
[SNIP]
(hope this doesn't wrap)

http://www1.us.dell.com/content/products/productdetails.aspx/pedge_600sc?c=u
s&cs=555&l=en&s=biz
it did ! (lol)

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[Asterisk-Users] Asterisk integrated with ventrilo or teamspeak

[e-smith]

Hi,
I would like to get info about integrating either Teamspeak or Ventrilo with
Asterisk.

Ventrilo and Teamspeak is free voice conferense servers/clients that are
commonly used for online voice conference over internet (IP).
Both has their own clients!

I have had some request about accessing them from telephone lines as well as
Internet and I would like to hear if anyone has done anything like it ?


Ref.
- Ventrilo www.ventrilo.com
- Teamspeak www.teamspeak.org


Kind Regards
Mats

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RE: [Asterisk-Users] PCI 3.3 V

[Juha Suhonen]

On Tue, 25 Nov 2003, Tom Walsh wrote:

> ::Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
> ::any motherboard with PCI 3.3 . Any sugestions!?

> Dell offers a tower that I know for certain has 3.3V PCI bus. PowerEdge
> 600SC.

Dell PowerEdge 1750 (1U rackmount with up to 2* Xeon DP) has a 3.3V PCI
bus, and TE410P seems to work just fine in it.


-- juhas
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Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)

[Sri]

unsubscribe

Robert G. Werner wrote:

The problem with -newbies (or even some PC name for it) is that people
won't use it.  

Rarely do people self select themselves as more ignorant than they
really are.  I'm afraid the "noob" problem just can't be resolved with
any structural changes.  

Personally,  I don't try to read the whole output of any list.  I look
for subjects that are interesting and then liberally use the delete
all option of my mail reader.  

I don't have any good suggestions for other lists/names,  though so
... I guess ignore this.
 



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Re: [Asterisk-Users] Outgoing-call and enter user in Conference - repost

[Andrew Thompson]

- Original Message - 
From: "Areski" <[EMAIL PROTECTED]>
To: "Asterisk-Users Mailing-list" <[EMAIL PROTECTED]>
Sent: Tuesday, November 25, 2003 12:13 PM
Subject: [Asterisk-Users] Outgoing-call and enter user in Conference - repost


> Hi all,
> 
> 
> Just wondering if someone have already done something like that :
> 
> 
> SIP Client_A ---> 1)call --->  ASTERISK  ---> 2)outgoingcall-PSTN-->Client_B
>  |
>  |
>3) Enter conference | 
>  MeetMe <'
>  with user A
> 
> 
> Make 2 user in conference (point 1 and 2), it's definitely easy, but call an other 
> user
> and put the both in conference,I still don't have any idea how to do it!
> 

I'm not speaking from experience, but couldn't you set up an extension for meetme, and 
just transfer your callers into it?

1) Make/take call.
2) transfer caller to meetme
3) dial another user
4) transfer to meetme
5) lather, rinse, repeat...


-
Andrew Thompson
NetResults, Inc.
(910) 215-9991 x301

Ë^®+$RÇ«²f¢–)à–+-Ë^®+$RÇ«²X¬¶Çb‚+¦r‰¡¶ÚþX¬¶Çb‚+¦r‰¿™¨¥™©ÿ–+-Šwèý«-z¸¬’ë®

Re: [Asterisk-Users] Prompt recording

[Chris Albertson]

--- Steve Underwood <[EMAIL PROTECTED]> wrote:
> Steven Critchfield wrote:
> 
> >On Tue, 2003-11-25 at 09:24, Jerimiah Cole wrote:
> >  
> >
> >>Does anybody have useful tips on creating good quality recordings
> for 
> >>use with prompts in asterisk?  I'm interested in hearing input on 
> >>hardware (mics, dats, sound cards, etc) and software (recording 
> >>software, dsp) as well as recording techniques.

I've found that a mic will pick up a computer fan noise if there is
a PC ruung inplave in the same room.  I was surprized to find out
just how bad a relitively "quiet" PC fan sounds.  I couldn't
figure it out at first as the noise on the recording did not
sound like a fan.

I deally I think you should use some kind of tape recorder and
then run the recording into the computer with a line-in connection
or maybe SPDIF from a DAT

Record at a "high" setting like 16-bit 48Khz and then scale
it down to the required format that way you use the full
dynamic range of the low-end 8x8 format.

You need "dead quiet" in the recording room not just "low noise"

You should be able to do as well as the sample voice recording
which as very good (unless you don't like the slight accent.)

=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

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[Asterisk-Users] prepaid application available?

[David Luyens]

Does anyone know of some people having developped a prepaid application
on asterisk?

David

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Re: [Asterisk-Users] Asterisk integrated with ventrilo or teamspeak

[Steven Critchfield]

Please read past rants about the action you took to create this message.
Hint: You broke the thread by replying to an unrelated thread.

On Tue, 2003-11-25 at 11:24, e-smith wrote:
> Hi,
> I would like to get info about integrating either Teamspeak or Ventrilo with
> Asterisk.
> 
> Ventrilo and Teamspeak is free voice conferense servers/clients that are
> commonly used for online voice conference over internet (IP).
> Both has their own clients!
> 
> I have had some request about accessing them from telephone lines as well as
> Internet and I would like to hear if anyone has done anything like it ?
> 
> 
> Ref.
> - Ventrilo www.ventrilo.com
> - Teamspeak www.teamspeak.org

And these accomplish what that asterisk doesn't do already?
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] prepaid application available?

[Steven Critchfield]

On Tue, 2003-11-25 at 11:57, David Luyens wrote:
> Does anyone know of some people having developped a prepaid application
> on asterisk?

Please use the Truly lazy way to start a new thread and click on the
mailing list address in a message so that the rest of us who have a clue
don't have to get annoyed at you.

Archives: Archives: Archives:
Please see the Archives. Granted the threads may be harder to find as
some are cloaked in secretive questions that barely give away what they
where developing.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Asterisk integrated with ventrilo or teamspeak

[Andrew Gillham]

Steven Critchfield wrote:

Please read past rants about the action you took to create this message.
Hint: You broke the thread by replying to an unrelated thread.
 

Could all of the thread police please just reply personally to the
offending party?
The amount of people interested in the rant is probably similar to the
amount interested in rants about grammar, spelling, newlines, etc.
-Andrew

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Re: [Asterisk-Users] Asterisk integrated with ventrilo or teamspeak

[Steven Critchfield]

On Tue, 2003-11-25 at 12:50, Andrew Gillham wrote:
> Steven Critchfield wrote:
> 
> >Please read past rants about the action you took to create this message.
> >Hint: You broke the thread by replying to an unrelated thread.
> >
> >  
> >
> Could all of the thread police please just reply personally to the
> offending party?
> 
> The amount of people interested in the rant is probably similar to the
> amount interested in rants about grammar, spelling, newlines, etc.

While I understand you not being interested in it, it is important to
remind as many people. You tend to speed less through a section of road
if you see the police lights on the side with cars pulled over. 

Also, I was trying to keep it brief so as to not be too annoying.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] AGI Rocks!! (A happy camper)

[WipeOut]

A note to all those who are avoiding writing up an AGI becasue it looks 
two complicated..

I have just written up my first and its awesome.. It makes Asterisk open 
to all sorts of possibilities.. let your imagination run wild..

I put off writing an AGI script because a) I could not find any docs b) 
it looked like the only way to do it was perl and I know nothing about 
perl and c) I am not a coder, more a simple sysadmin..

Eventually I decided to give it a try, and seeing as the language I know 
the best is PHP I decided to do my script in PHP.. The truth of the 
matter is that once you have worked out how to get your variables from 
Asterisk into a usable array or whatever you are happy using the rest is 
a piece of cake..

It goes without saying that what I have done is stupidly simple but for 
a non coder its still a giant achivement..

So all I can say is thanks to the Mark and the team for dreaming up 
AGI... To those who are putting off giving it a go, don't!!.. Dive in 
and give it a try..

PS. I won't be much help to anyone on AGI related questions, the list is 
still you best bet, this was just to hopefully inspire others to give it 
a try..

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RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)

[Steven Sokol]

Billy,

Thanks for the observations.  The caller ID on outbound is unfortunately
part of the way the Manager interface does outbound calling.  Since it
establishes a call from Asterisk to YOU, then establishes another call
from Asterisk to CALLED PARTY then bridges YOU and CALLED PARTY, you are
not actually the "caller" in the call to VM.

The Hold issue is also one I have yet to find my way around.  The
manager interface doesn't have a "Hold" method (yet).  I may very well
hack one into the manager code so that you can Hold and Reconnect calls,
(plus know when your call has been held/reconnected).

Another solution to this for SIP is to use the REFER methods to direct
the UA (in your case the GS Budgetone).  I don't know which UAs support
this.  Considering the limits of the GS phone, I doubt it would.

I am working on a new version of Call Manager and I will have it ready
sometime after the Thanksgiving holiday.

Thanks,

Steve

> -Original Message-
> From: Billy Huddleston [mailto:[EMAIL PROTECTED]
> Sent: Monday, November 24, 2003 12:45 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
> (Alpha)
> 
> Been trying out your call manager.. Nice work!  Have a few
opservations to
> tell you...
> 
> When dialing out, it doesn't do callerid, I found this out, because I
use
> callerid to know what extension has called voicemail, so all they have
to
> enter in is thier pass code.
> 
> Hold doesn't work with GS Budgetone at all, and, I'm not exactly sure
what
> the deal is with transfer, I can't get it to work..
> 
> Thanks, Billy
> 
> 
> 
> - Original Message -
> From: "Steven Sokol" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Saturday, November 22, 2003 11:21 AM
> Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
> (Alpha)
> 
> 
> > Zoa,
> >
> > When the boxes are red, that usually indicates that the channel is
busy.
> > In the screen shots I sent earlier you will see that one of the
buttons
> > is red:
> >
> > http://www.sokol-associates.com/images/AstMgr.jpg
> >
> > Notice that only the station marked "Test Xten" is red.  This
station is
> > busy (on another call).  I don't know if that has anything to do
with
> > your issue, but I thought I would throw that out.
> >
> > The message you reference below is a "Status" message.  In this
program
> > the Status messages really only serve as keep-alives.  Every 30
seconds
> > the system issues a command "Action:Status" to keep NATs from
closing
> > the connection due to lack of traffic.
> >
> > Try this:  open the command window and try manually executing some
of
> > the CLI commands.  Try "sip show peers" to make sure the SIP peers
are
> > registered.  Also try "sip show channels" to see if there is already
a
> > call terminated at the channel you are calling.
> >
> > I will try to diagnose this further if you can send some additional
> > information.  Please include the monitor.conf file, and if possible
a
> > - trace from Asterisk.
> >
> > Thanks,
> >
> > Steve
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of zoa
> > > Sent: Saturday, November 22, 2003 4:13 AM
> > > To: [EMAIL PROTECTED]
> > > Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows
0.0.1
> > > (Alpha)
> > >
> > > [201]
> > > Username=davy
> > > Technology=SIP
> > > DeviceID=davy
> > >
> > > [202]
> > > Username=pieter
> > > Technology=SIP
> > > DeviceID=pieter
> > >
> > > 201 is the extension from in extensions.conf
> > > davy = the thing between brackets in sip.conf
> > >
> > > When i try to click on one of the red boxes in the manager, i
always
> > get:
> > >
> > > Event: Status -|- Channel: SIP/davy-1a07 -|- CallerID: davy -|-
State:
> > Up
> > > -|- Context: sip -|- Extension: 202 -|- Priority: 1 -|- Link:
> > > SIP/pieter-e582 -|- Uniqueid: 1069581.102
> > > Response: Error -|- Message: Invalid channel
> > > At 21:06 21/11/2003 -0600, you wrote:
> > > >Here's the structure for the monitor.conf file:
> > > >
> > > >[1101]  Extension Number (from extensions.conf in
> > > >Asterisk)
> > > >UserName=Blah Blah  Label.  Simply sets the caption for the
> > button.
> > > >Technology=SIP  Technology used for stations (SIP, MGCP,
Zap,
> > > >etc.)
> > > >DeviceID=1101   Device identifier (from sip.conf in this
> > case)
> > > >
> > > >All of the Technology values are normal asterisk values except
for
> > APP,
> > > >which is an application (like Voicemail or MOH or MeetMe) and
PSTN,
> > > >which is a number outside of the Asterisk inside dial plan.
> > > >
> > > >I hope this helps.  Remember that for PSTN and APP values, the
> > bracketed
> > > >Extension number and the DeviceID need to be the same.
> > > >
> > > >Regards,
> > > >
> > > >Steve
> > > >
> > > > > -Original Message-
> > > > > From: [EMAIL PROTECTED]
> > [mailto:asterisk-users-
> > > > > [EMAIL PROTECTED] On Beh

[Asterisk-Users] ADSI Programming - Found Guide on Designing Apps

[Jonathan Biggs]

Found some info on the Web that may help some 
of the ADSI programmers out there.

The following guide is for a WebSphere implementation
but the average developer type should be able to pull
enough out of it to help writing ADSI scripts for
Asterisk.  Seemed to have good overview of ADSI
capabilities and a lot of direct correlation to the
asterisk ADSI scripts 

I did not read all 213 pages yet

WebSphere Voice Response for AIX V3.1 Programming for
the ADSI feature (SC34-5380-04)

Abstract:
WebSphere Voice Response for AIX V3.1 Programming for
the ADSI feature.
This book tells you how to install the Analog Display
Services Interface (ADSI) component of WebSphere Voice
Response for AIX. It also tells you how to design
voice applications that run on ADSI telephones.

http://www.elink.ibmlink.ibm.com/public/applications/publications/cgibin/pbi.cgi?CTY=US&FNC=SRX&PBL=SC34-5380-04

Good luck.. Hope this helps


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Re: chan_h323 vs asterisk-oh323 (Was Re: [Asterisk-Users] Cisco to asterisk termination with h323 and g729 finally works.)

[Richard Lyman]

see below

Michael Manousos wrote:
Max Tulyev wrote:

В сообщении от 24 Ноябрь 2003 21:27 Jeremy McNamara написал:


I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.


I have asterisk-oh323-0.5.7.tar.gz and * from CVS @ 20 Nov 2003.
PWLib 1.5.2, OpenH323 1.12.2
ATA-186(h.323)->gnugk->*->7940(SIP)

I see segmentation fault on asterisk when RTP starts. I don't know 
why, I'll 


If you don't provide the needed info the problem
will remain.
this is a digium list, and this thread started from jeremy asking 
about info/diffs so that a doc can be created to HELP digium 
users.  this is NOT a defend your product soapbox.

please try and control yourself.

thanks

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Re: [Asterisk-Users] ADSI Programming - Found Guide on Designing Apps

[Jonathan Biggs]

Book costs $49.50

PDF Download - free

--- Jonathan Biggs <[EMAIL PROTECTED]> wrote:
> 
> Found some info on the Web that may help some 
> of the ADSI programmers out there.
> 
> The following guide is for a WebSphere
> implementation
> but the average developer type should be able to
> pull
> enough out of it to help writing ADSI scripts for
> Asterisk.  Seemed to have good overview of ADSI
> capabilities and a lot of direct correlation to the
> asterisk ADSI scripts 
> 
> I did not read all 213 pages yet
> 
> WebSphere Voice Response for AIX V3.1 Programming
> for
> the ADSI feature (SC34-5380-04)
> 
> Abstract:
> WebSphere Voice Response for AIX V3.1 Programming
> for
> the ADSI feature.
> This book tells you how to install the Analog
> Display
> Services Interface (ADSI) component of WebSphere
> Voice
> Response for AIX. It also tells you how to design
> voice applications that run on ADSI telephones.
> 
>
http://www.elink.ibmlink.ibm.com/public/applications/publications/cgibin/pbi.cgi?CTY=US&FNC=SRX&PBL=SC34-5380-04
> 
> Good luck.. Hope this helps
> 
> 
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>
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[Asterisk-Users] Options for 3rd party call control

[Alistair Cunningham]

I am working on a project on 3rd party call control for a call center, for
which I think Asterisk may be useful. What I would like to do is:

- Have a call come in to Asterisk.

- Asterisk asks another machine, over a slow IP link, such as a modem, how it
  should route the call. Asterisk passes the called and calling numbers.

- This other machine looks up the destination, based on called and calling
  numbers, in an SQL database, and responds to Asterisk.

- When Asterisk gets a reply, it routes the call.

- During the call, this other machine may ask Asterisk to re-route the call.

- If this happens, Asterisk hangs up on the party it called, plays a 'please
  hold' message to the caller, then connects them to the new destination.

- All this happens over a mixture of basic rate ISDN, analogue lines, H.323,
  and SIP.
  
Looking at the Asterisk website, I don't see any options to do this. Has
anyone given this any thought? Would such a thing be difficult to write?

-- 
Alistair Cunningham,
Email: [EMAIL PROTECTED]
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Re: [Asterisk-Users] cdr_unixodbc

[Philipp von Klitzing]

Hi!

Indeed great move, Brian!

>   What i'm trying to figure out is how to keep the sql statement
> globally prepaired then just call SQLExecute but the docs for all this are
> hard to come by.  I really can't tell much diffrence in odbc over mysql in
> speed but I don't have a bazillion calls going thru at once.  It does add
> a layer of abstraction that should allow it to work with just bout any DB
> server that has a driver for unixODBC

Ok, next dream: Have new applications "DBputODBC", "DBdelODBC" etc?

Philipp


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Re: [Asterisk-Users] AGI Rocks!! (A happy camper)

[Andrew Thompson]

- Original Message - 
From: "WipeOut" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 25, 2003 2:14 PM
Subject: [Asterisk-Users] AGI Rocks!! (A happy camper)


> A note to all those who are avoiding writing up an AGI becasue it looks 
> two complicated..
> 
> I have just written up my first and its awesome.. It makes Asterisk open 
> to all sorts of possibilities.. let your imagination run wild..
> 
> I put off writing an AGI script because a) I could not find any docs b) 
> it looked like the only way to do it was perl and I know nothing about 
> perl and c) I am not a coder, more a simple sysadmin..
> 
> Eventually I decided to give it a try, and seeing as the language I know 
> the best is PHP I decided to do my script in PHP.. The truth of the 
> matter is that once you have worked out how to get your variables from 
> Asterisk into a usable array or whatever you are happy using the rest is 
> a piece of cake..
> 
> It goes without saying that what I have done is stupidly simple but for 
> a non coder its still a giant achivement..
> 
> So all I can say is thanks to the Mark and the team for dreaming up 
> AGI... To those who are putting off giving it a go, don't!!.. Dive in 
> and give it a try..
> 

I believe I read Mark or someone say many of the builtin apps would 
become AGI scripts in the future. 

Given how many people cry over the voicemail layout, I've been thinking 
about writing an AGI script for voicemail. PHP would be my first choice,
unless I can find my perl book.

Maybe I will go hack something out...

Has anyone already started work on a to be released script like this?

-
Andrew Thompson,µêâ²E,z»&j)bž  
b²Ð,µêâ²E,z»%ŠËlv("ºg(šm§ÿåŠËlv("ºg(›ùšŠYšŸùb²Ø§~Ú²×«ŠÉ.±êì

Re: [Asterisk-Users] AGI Rocks!! (A happy camper)

[costas]

I was just looking at AGI with PHP myself. I just have a real dumb question. How does 
Linux know to send $stdout(or echo) to *? What if there are other apps open as well 
waiting for input. WOn't they get the output?

Also, how does the AGI know to read from $stdin is * input?

Costas

-- Original Message --
From: WipeOut <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
Date:  Tue, 25 Nov 2003 19:14:48 +

>A note to all those who are avoiding writing up an AGI becasue it looks 
>two complicated..
>
>I have just written up my first and its awesome.. It makes Asterisk open 
>to all sorts of possibilities.. let your imagination run wild..
>
>I put off writing an AGI script because a) I could not find any docs b) 
>it looked like the only way to do it was perl and I know nothing about 
>perl and c) I am not a coder, more a simple sysadmin..
>
>Eventually I decided to give it a try, and seeing as the language I know 
>the best is PHP I decided to do my script in PHP.. The truth of the 
>matter is that once you have worked out how to get your variables from 
>Asterisk into a usable array or whatever you are happy using the rest is 
>a piece of cake..
>
>It goes without saying that what I have done is stupidly simple but for 
>a non coder its still a giant achivement..
>
>So all I can say is thanks to the Mark and the team for dreaming up 
>AGI... To those who are putting off giving it a go, don't!!.. Dive in 
>and give it a try..
>
>PS. I won't be much help to anyone on AGI related questions, the list is 
>still you best bet, this was just to hopefully inspire others to give it 
>a try..
>
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Re: [Asterisk-Users] Asterisk integrated with ventrilo or teamspeak

[Philipp von Klitzing]

Hi!

> > - Ventrilo www.ventrilo.com
> > - Teamspeak www.teamspeak.org
> 
> And these accomplish what that asterisk doesn't do already?

Concerning teamspeak:

- graphical client where a channel or sub-channel moderator can perform 
all kinds of action like moving users to a different channel, mute them, 
remove speaking rights, grant channel moderator rights and a ton of 
similar things like ban, kick, whisper to individuals etc

- integrated text chat (limited in features, but still very useful)

- mainly designed for Internet team games, but due to that also great for 
bigger conferences/ meetings

Cheers, Philipp


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Re: [Asterisk-Users] Asterisk integrated with ventrilo or teamspeak

[Philipp von Klitzing]

Hi!

> I would like to get info about integrating either Teamspeak or Ventrilo with
> Asterisk.

For conferencing Teamspeak is great, no doubt. Being able to join TSS 
conferences from * would be a great thing indeed. Maybe this is a 
suggestion worth to post on the TSS forum as well?

Cheers, Philipp


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Re: [Asterisk-Users] Options for 3rd party call control

[Steven Critchfield]

On Tue, 2003-11-25 at 13:51, Alistair Cunningham wrote:
> I am working on a project on 3rd party call control for a call center, for
> which I think Asterisk may be useful. What I would like to do is:
> 
> - Have a call come in to Asterisk.
> 
> - Asterisk asks another machine, over a slow IP link, such as a modem, how it
>   should route the call. Asterisk passes the called and calling numbers.
> 
> - This other machine looks up the destination, based on called and calling
>   numbers, in an SQL database, and responds to Asterisk.
> 
> - When Asterisk gets a reply, it routes the call.
> 
> - During the call, this other machine may ask Asterisk to re-route the call.
> 
> - If this happens, Asterisk hangs up on the party it called, plays a 'please
>   hold' message to the caller, then connects them to the new destination.
> 
> - All this happens over a mixture of basic rate ISDN, analogue lines, H.323,
>   and SIP.
>   
> Looking at the Asterisk website, I don't see any options to do this. Has
> anyone given this any thought? Would such a thing be difficult to write?

Why go out over a slow link to ask how to route when you could do any
number of lookups locally on the machine. You could write your lookup
logic into the dialplan. You could write your lookup logic into an AGI
script. You could do just about any lookups possible without the need to
go remote, but if you had to go remote, then thats easy too.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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