date:20060920

Hi all,
trixbox has taken control of my asterisk system, i dont like that. i just installed trixbox for rersearch purpose now i want to uninstall it and do some research on asterisk. So plz tell me how to uninstall trixbox. will it uninstall asterisk also?
-- RegardsRizwan HishamSoftware Engineer 
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[asterisk-users] Re: mpg123

2006-09-20 Thread Tomislav Parčina

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 Hi all,
 I'm using * 1.0.9 which use mpg123 for music on hold. But sometimes
 starts eating up a lot of CPU.
 I sthere any alternative method to use moh without use mpg123?
 I tryied this http://astrecipes.net/?n=152 but i doesn't wotks for me.
  
 Anyone can help me pls ?

Upgrade to Asterisk 1.2 and use native sounds.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Uninstalling Trixbox

2006-09-20 Thread Sharon Lim

>From my understanding, tribox is to control asterisk via web interface. so, if u want to uninstall the tribox, i guess just delete the web folder will do then do can edit direction frm your asterisk files. 
On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
trixbox has taken control of my asterisk system, i dont like that. i just installed trixbox for rersearch purpose now i want to uninstall it and do some research on asterisk. So plz tell me how to uninstall trixbox. will it uninstall asterisk also?
-- RegardsRizwan HishamSoftware Engineer 

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http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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RE: [asterisk-users] SIP Lines Example Citel

2006-09-20 Thread Steve Langstaff

The following works for me, for e.g. extension number '5301' with a
secret of 'secret' on server '192.168.1.1'

Addressing
SIP Address-of-Record (AOR): sip:[EMAIL PROTECTED]

Registrar Server
Domain: asterisk
Expiration: 3600

Authorisation
Update Authorisation: checked
Username: 5301
Realm: asterisk
Password: secret
Retype Password: secret

If you have any more questions, mailto:[EMAIL PROTECTED] will be happy
to help.

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Steve Totaro
 Sent: 19 September 2006 18:20
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] SIP Lines Example Citel
 
 Anyone know how to setup the SIP lines on a Citel box so it 
 can register with Asterisk.  I keep getting Unauthorized 
 and I have tried every different combination of settings that 
 I can think of.  I am not sure what fields are required or 
 what information goes where in the Citel interface.
 
 Thanks,
 Steve Totaro
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RE: [asterisk-users] Alcatel OXO Sip

2006-09-20 Thread Christian Gatti

No, it says I don't know what to do if the via is *not* SIP/2.0/UDP
Yes, my error. 

I tested with asterisk 1.2.10 and now it works.
Reason: Steve Langstaff found that in my old * version there was case
sensitive strcmp which caused the problem.

Thanks to you all,
Christian

 Subject: RE: [asterisk-users] Alcatel OXO Sip
 
 Yeah, I've just downloaded the source for 1.0.7, and found the
 following:
 
   if (strcmp(via, SIP/2.0/UDP)) {
   ast_log(LOG_WARNING, Don't know how to respond via
'%s'\n, via);
   return -1;
   }
 
 So it's a case-sensitive compare on the version you are running!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Candler
Sent: Tuesday, September 19, 2006 18:51 PM
To: Christian Gatti
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Alcatel OXO Sip

On Tue, Sep 19, 2006 at 02:53:17PM +0200, Christian Gatti wrote:
 It the question why does asterisk has problems with SIP/2.0/udp or 
 SIP/2.0/UDP
 
 if (strcasecmp(via, SIP/2.0/UDP)) {
   ast_log(LOG_WARNING, Don't know how to respond via '%s'\n, via);
   return -1;
 }
 
 This code says: I don't know what to do with a SIP/2.0/UDP in a via 
 and blocks (return -1).

No, it says I don't know what to do if the via is *not* SIP/2.0/UDP
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Re: [asterisk-users] grandstream gxp 2000 does not display names when calling out

2006-09-20 Thread Michiel van Baak

On 23:47, Tue 19 Sep 06, Michael Neuhauser wrote:
 On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote:
  michael,
  at my real job, the phones display peoples names when calling out from
  your phone. how is this done?

Maybe they put the names in the phones internal addressbook
?
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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Re: [asterisk-users] Uninstalling Trixbox

On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote:
 Hi all,
 trixbox has taken control of my asterisk system, i dont like that. i just
 installed trixbox for rersearch purpose now i want to uninstall it and do
 some research on asterisk. So plz tell me how to uninstall trixbox. will it
 uninstall asterisk also?

Trixbox has not taken over your system. It is your system. It is not
just the web interface. Trixbox is a customized CentOS distribution. It
uses a number of its own packages in its own yum source (in addition to
standard CentOS packages), and has a number of non-default settings.
e.g: selinux disabled.

Or you may be confusing it with FreePBX?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] stress a server with a tool

2006-09-20 Thread nik600


hi

is there any software usable to simulate work on an asterisk server?

I'm interested in it to evaluate the level of currently calls that a
server can support
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Re: [asterisk-users] mpg123

On Tue, Sep 19, 2006 at 06:53:01PM +0200, Giordano Grandis wrote:
 Hi all,
 I'm using * 1.0.9 which use mpg123 for music on hold. But sometimes
 starts eating up a lot of CPU.
 I sthere any alternative method to use moh without use mpg123?
 I tryied this http://astrecipes.net/?n=152 but i doesn't wotks for me.

Those instructions are for Asterisk 1.2 . 

In 1.0.x the MoH player will look for *.mp3 files. Xorcom Rapid has a
makefile that convers mp3 files to wav files with the extension .wav.mp3 .
Quite similar o the examples in the abover case, only a different output
file name.

Then you need to follow the sample musiconhold.conf file and define a
custom method with a script that will play those wav files to the
standard output.

If the out format was signed linear rather than wav, you could basically
use cat .

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] BRI: Asterisk disconnecting on 'call diverted' message?

2006-09-20 Thread Benoit Panizzon

Hi All

I'm tracing a strange BRI Q.931 Problem with Asterisk 1.2.4.

I call a number which is diverted to another number.
Asterisk seams to take this divertification message as a hangup message:

BRI Trace:

-- Executing Dial(IAX2/magma-1, Zap/g7/0418103734|90) in new stack
2 -- Making new call for cr 134
-- Requested transfer capability: 0x00 - SPEECH
2  Protocol Discriminator: Q.931 (8)  len=38
2  Call Ref: len= 1 (reference 6/0x6) (Originator)
2  Message type: SETUP (5)
2  [2 042  032  802  902  a32 ]
2  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
2   Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
(16)
2   Ext: 1  User information layer 1: A-Law (35)
2  [2 182  012  812 ]
2  Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Preferred 
Dchan: 0
2 ChanSel: B1 channel
2  ]
2  [2 6c2  0c2  412  812  302  362  312  382  312  312  352  372  312  312 ]
2  Calling Number (len=14) [ Ext: 0  TON: Subscriber Number (4)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
2Presentation: Presentation permitted, user 
number passed network screening (1) '06181157**' ]
2  [2 702  0a2  a12  342  312  382  312  302  332  372  332  342 ]
2  Called Number (len=12) [ Ext: 1  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '418103734' ]
-- Called g7/0418103734
2  Protocol Discriminator: Q.931 (8)  len=7
2  Call Ref: len= 1 (reference 134/0x86) (Terminator)
2  Message type: SETUP ACKNOWLEDGE (13)
2  [2 182  012  892 ]
2  Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Exclusive 
Dchan: 0
2 ChanSel: B1 channel
2  ]
2 -- Processing IE 24 (cs0, Channel Identification)
2  Protocol Discriminator: Q.931 (8)  len=7
2  Call Ref: len= 1 (reference 134/0x86) (Terminator)
2  Message type: CALL PROCEEDING (2)
2  [2 272  012  fb2 ]
2  Notification indicator (len= 3): Ext: 1  Call is diverting (123)

Why is the number being diverted to not advertized? On the SS7 Trunk the 
number is presented and I see that presentation is allowed.

2 -- Processing IE 39 (cs0, Notification Indicator)
-- Zap/4-1 is proceeding passing it to IAX2/magma-1
2  Protocol Discriminator: Q.931 (8)  len=57
2  Call Ref: len= 1 (reference 134/0x86) (Terminator)
2  Message type: DISCONNECT (69)

Why this disconnect? If I connect a ISDN Phone directly to the BRI I don't get 
disconnected but get the message from the telco that the destination is 
unreachable at the moment. Why is this audio not passed to the caller?

2  [2 082  022  8a2  9b2 ]
2  Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: 
Network beyond the interworking point (10)
2   Ext: 1  Cause: Unknown (27), class = Normal Event (1) ]
2  [2 1c2  1d2  912  a12  1a2  022  022  722  b92  022  012  232  302  112  
302  0f2  a12  0d2  812  032  462  522  2e2  a22  062  812  012  002  822  
012  012 ]
2  Facility (len=31, codeset=0) [ 2 0x91, 0xa1, 0x1a, 0x02, 0x02, 'r', 0xb9, 
0x02, 0x01, 0x23, '0', 0x11, '0', 0x0f, 0xa1, 0x0d, 0x81, 0x03, 'FR', 0x2e, 
0xa2, 0x06, 0x81, 0x01, 0x00, 0x82, 0x01, 0x012  ]
2  [2 1e2  022  8a2  882 ]
2  Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 
0   Location: Network beyond the interworking point (10)
2Ext: 1  Progress Description: Inband 
information or appropriate pattern now available. (8) ]
2  [2 1e2  022  8a2  822 ]
2  Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 
0   Location: Network beyond the interworking point (10)
2Ext: 1  Progress Description: Called 
equipment is non-ISDN. (2) ]
2  [2 282  082  462  522  2e2  202  302  2e2  302  302 ]
2  Display (len= 8) [ FR. 0.00 ]
2 -- Processing IE 8 (cs0, Cause)
2 -- Processing IE 28 (cs0, Facility)
2 Handle Q.932 ROSE Invoke component
2 -- Processing IE 30 (cs0, Progress Indicator)
2 -- Processing IE 30 (cs0, Progress Indicator)
2 -- Processing IE 40 (cs0, Display)
-- Channel 0/1, span 2 got hangup request
-- Zap/4-1 is circuit-busy
2 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, 
peerstate Disconnect Request
2  Protocol Discriminator: Q.931 (8)  len=8
2  Call Ref: len= 1 (reference 6/0x6) (Originator)
2  Message type: RELEASE (77)
2  [2 082  022  812  9b2 ]
2  Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: 
Private network serving the local user (1)
2   Ext: 1  Cause: Unknown (27), class = Normal Event (1) ]
-- Hungup 'Zap/4-1'


Mit freundlichen Grüssen

Benoit Panizzon
-- 
I m p r o W a r e   A G-System Services
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01

Re: [asterisk-users] Asterisk AGI question

2006-09-20 Thread Yoann Aubineau

Le mardi 19 septembre 2006 à 15:30 -0500, David R. a écrit :
 Can AGI be used to have a web application talk back and forth between
 Asterisk and itself?  What if the web application is on a separate
 box? 

As Stefan Reuter previously stated, there's no problem running your AGI
application remotely. It's extremely fast (hence the name FastAGI) and
it also frees the Asterisk box from any load your application could
produce (think of calculation or database query processing)

However, even though AGI is a lot like CGI for Asterisk, it doesn't mean
you can use web applications as AGI applications. Ok, you could tell
Asterisk what to do. But how would you get responses back from
Asterisk? 

Or maybe you've got a genius idea I couldn't think of. In that case, let
us know!

Yoann

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Re: [asterisk-users] Uninstalling Trixbox

no, im not using FreePBX, actually freepbx is a part of trixbox as is
sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE.
By saying that 'It has taken control of my system', i meant asterisk.
Now i dont want any web based interface to asterisk. i only want
asterisk on my system. So plz help me uninstall trixbox.

And Sharon, thanx for the tip, but what about the rest of the scripts
trixbox has installed on my system. for example i dont want to start
asterisk on system startup, but trixbox does that. so anymore help will
be helpfull :)On 9/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote: Hi all, trixbox has taken control of my asterisk system, i dont like that. i just installed trixbox for rersearch purpose now i want to uninstall it and do
some research on asterisk. So plz tell me how to uninstall trixbox. will it uninstall asterisk also?Trixbox has not taken over your system. It is your system. It is notjust the web interface. Trixbox is a customized CentOS distribution. It
uses a number of its own packages in its own yum source (in addition tostandard CentOS packages), and has a number of non-default settings.e.g: selinux disabled.Or you may be confusing it with FreePBX?
--Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755iax:[EMAIL PROTECTED]
+972-50-7952406jabber:[EMAIL PROTECTED][EMAIL PROTECTED] http://www.xorcom.com
___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- RegardsRizwan HishamSoftware Engineer
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Re: [asterisk-users] Uninstalling Trixbox

2006-09-20 Thread Sharon Lim

hehehe, then you have to edit the startup manual. I dont think so there is a way to uninstall the tribox. for example to disable start up for asterisk, i think you can try type the command chkconfig asterisk stop ..good luck
On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
no, im not using FreePBX, actually freepbx is a part of trixbox as is
sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE.
By saying that 'It has taken control of my system', i meant asterisk.
Now i dont want any web based interface to asterisk. i only want
asterisk on my system. So plz help me uninstall trixbox.

And Sharon, thanx for the tip, but what about the rest of the scripts
trixbox has installed on my system. for example i dont want to start
asterisk on system startup, but trixbox does that. so anymore help will
be helpfull :)On 9/20/06, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote: Hi all, trixbox has taken control of my asterisk system, i dont like that. i just installed trixbox for rersearch purpose now i want to uninstall it and do
some research on asterisk. So plz tell me how to uninstall trixbox. will it uninstall asterisk also?Trixbox has not taken over your system. It is your system. It is notjust the web interface. Trixbox is a customized CentOS distribution. It
uses a number of its own packages in its own yum source (in addition tostandard CentOS packages), and has a number of non-default settings.e.g: selinux disabled.Or you may be confusing it with FreePBX?
--Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755
iax:[EMAIL PROTECTED]
+972-50-7952406jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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-- RegardsRizwan HishamSoftware Engineer

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--asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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[asterisk-users] Channel kept busy when creating ssh tunnel via AGI

2006-09-20 Thread Giorgio Incantalupo


Hi,
I have a problem with Asterisk AGI command.
I wrote a script which  launches a shell command.
If I launch a normal command for example like ll  /tmp/tmp.txt, the 
AGI command launches the shell commands and then exits.


The problem is when I launch THIS command to create an ssh tunnel in 
background:

*ssh -f -N -l asterisk -R 2050:localhost:22 192.168.0.1*

The tunnel command above works well if launched via shell but if I 
launch it using the AGI script, it opens the tunnel but leaves a (SIP or 
ZAP) channel  in use (I checked it typing SIP/ZAP SHOW CHANNELS).
The channel closes only when I kill the tunnel process. After killing 
the process Asterisk console shows:


-- AGI Script tunnel.py completed, returning 0

Is there anybody who knows why the channel remains busy?

TIA


Giorgio Incantalupo
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[asterisk-users] agi

2006-09-20 Thread Gopal krishnan

hi asterisk' ians   How to write agi scripts, how to see the output..  thanks in advance...  
	

	
		Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.
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Re: [asterisk-users] stress a server with a tool

2006-09-20 Thread Charles Wang


The radvision's prolabs is your best choice for SIP or H.323.

2006/9/20, nik600 [EMAIL PROTECTED]:

hi

is there any software usable to simulate work on an asterisk server?

I'm interested in it to evaluate the level of currently calls that a
server can support
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--

Best Regards
Charles
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Re: [asterisk-users] Channel kept busy when creating ssh tunnel via AGI

2006-09-20 Thread BJ Weschke


On 9/20/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:

Hi,
I have a problem with Asterisk AGI command.
I wrote a script which  launches a shell command.
If I launch a normal command for example like ll  /tmp/tmp.txt, the
AGI command launches the shell commands and then exits.

The problem is when I launch THIS command to create an ssh tunnel in
background:
*ssh -f -N -l asterisk -R 2050:localhost:22 192.168.0.1*

The tunnel command above works well if launched via shell but if I
launch it using the AGI script, it opens the tunnel but leaves a (SIP or
ZAP) channel  in use (I checked it typing SIP/ZAP SHOW CHANNELS).
The channel closes only when I kill the tunnel process. After killing
the process Asterisk console shows:

-- AGI Script tunnel.py completed, returning 0

Is there anybody who knows why the channel remains busy?



The intent of an AGI script is to have a script/executable that
interacts with the channel. As such, the channel will hang around
waiting for input from, providing feedback to, and the eventual
completion of such script/executable.

If you don't want such behavior, you might want to take a look at
monitoring a specific channel event in the Asterisk manager and then
starting off your script upon the receipt of such an event through the
manager.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] Uninstalling Trixbox

On Wed, Sep 20, 2006 at 02:53:04PM +0500, Rizwan Hisham wrote:
 no, im not using FreePBX, actually freepbx is a part of trixbox as is
 sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE. By
 saying that 'It has taken control of my system', i meant asterisk. Now i
 dont want any web based interface to asterisk. i only want asterisk on my
 system. So plz help me uninstall trixbox.
 
 And Sharon, thanx for the tip, but what about the rest of the scripts
 trixbox has installed on my system. for example i dont want to start
 asterisk on system startup, but trixbox does that. so anymore help will be
 helpfull :)

Install a new system on a different machine and copy the bits you like. 
This will be simpler. 

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Mediant 1000

2006-09-20 Thread Rajkumar S


Hi,

I am looking for some docs to help configure a AudioCodes Mediant 1000
with asterisk, any  tips or examples are appreciated.

raj
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Re: [asterisk-users] SkypeOut with Asterisk?

2006-09-20 Thread Devraj Mukherjee


Thanks Sharon.

On 9/20/06, Sharon Lim [EMAIL PROTECTED] wrote:

I have successful link skype with asterisk with
http://www.nch.com.au/skypetosip/index.html but not sure
whether you need this.

here is another link
http://www.voip-info.org/wiki/index.php?page=Skype%20Gateways.

Good luck!



On 9/20/06, Devraj Mukherjee  [EMAIL PROTECTED] wrote:

Has anyone managed to use SkypeOut as your VoIP provider?

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Incredibles)
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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr

 good stuff mate.
 
 a few clarifications:
 you had static extensions.conf, realtime sipusers, etc, right?
 
 Also, abt features like call fwding, etc, which one is better,
 performance wise, using a mysql db, or use Asterisk's internal
 DB(berkeley db, isnt it?using those DBput n DBget ops)??Anyone's got any
 figures for these?
 
 This somewot spoils the fun in Asterisk, when talking of performance, to
 query the DB for every call . Sort of pulls things down. Any comments or
 observations guys?
 
 - Ben.

Ben,

Yes, static extensions.conf, realtime everything else. A realtime
dialplan never made much sense to me, as the dialplan shouldn't (in my
humble opinion) be that fluid anyway, it should be fairly static.

In terms of spoiling the fun and/or performance issues, let me note that
in my current implementation we not only have options being queried but
also realtime billing, permissions, limits, and carrier/trunk
performance data, all being pulled and calculated via the database. I
also have handy little timers returning the length of time it takes to
do the processing from request receipt to dial, and I'm still currently
under 1-2 seconds for entire call preparation including all the logic
that goes along with checking all features, the current account's
account status, balance and limits, AND all parent accounts in it's
billing chain. I haven't done a head to head with the berkley DB, but
I think part of the reason it's so fast is due to the highly normalized
database structure, which allows for efficient query design. It's not
all third form, but almost there :D.

I'm in the last days of ALPHA now with my current project. Once we
launch BETA, which will be a semi-public testing by invitation (Murph,
you still going to participate?), I should be able to find a few minutes
to outline the design.

One other quick thing, the berkley DB doesn't allow for clustering
either, MySQL does. Very nice to have your database distributed across
multiple nodes, makes for an easier time designing the failovers :D

Cheers,
Sherwood

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr

  I would like to know how you got Asterisk to function with 2500 SIP
 registrations.  Did you have qualify enabled?

Yes, qualify was enabled, using the standard length of qualification
period between checks. Very few accounts had custom qualify settings.

 What about the 500 simultaneous calls?  How many SQL hits were you
 doing (all said and done).  Any performance logs from the SQL server?
 
 I can't believe you got all this running on one box!

You have to remember, 500 simultaneous calls is not the same as
something like 20 calls per second. some of those calls may have been
quite long, and once the call's been placed, there's no database work
being done until the call ends.

I wish I had statistics from that setup, but I don't, we spent so much
time implementing new features and chasing down problems caused by using
a pre-RTA version of Asterisk with a patched in RTA setup.



-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: Re: [asterisk-users] Uninstalling Trixbox

2006-09-20 Thread Mike Dent


When you installed Trixbox did you not boot from the Trixbox install CD?
This installs CentOS and Trixbox.
I'm curious how you installed Trixbox?

Mike


On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote:

no, im not using FreePBX, actually freepbx is a part of trixbox as is
sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE. By
saying that 'It has taken control of my system', i meant asterisk. Now i
dont want any web based interface to asterisk. i only want asterisk on my
system. So plz help me uninstall trixbox.

 And Sharon, thanx for the tip, but what about the rest of the scripts
trixbox has installed on my system. for example i dont want to start
asterisk on system startup, but trixbox does that. so anymore help will be
helpfull :)


On 9/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
 On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote:
  Hi all,
  trixbox has taken control of my asterisk system, i dont like that. i
just
  installed trixbox for rersearch purpose now i want to uninstall it and
do
  some research on asterisk. So plz tell me how to uninstall trixbox. will
it
  uninstall asterisk also?

 Trixbox has not taken over your system. It is your system. It is not
 just the web interface. Trixbox is a customized CentOS distribution. It
 uses a number of its own packages in its own yum source (in addition to
 standard CentOS packages), and has a number of non-default settings.
 e.g: selinux disabled.

 Or you may be confusing it with FreePBX?

 --
 Tzafrir Cohen sip:[EMAIL PROTECTED]
 icq#16849755  iax:[EMAIL PROTECTED]
 +972-50-7952406  jabber:[EMAIL PROTECTED]
 [EMAIL PROTECTED] http://www.xorcom.com
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Rizwan Hisham
Software Engineer
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[asterisk-users] Registration doubt

2006-09-20 Thread god

Sir,
 I installed asterix in some system say with ip 172.16.7.63.From some other windows system say 172.16.7.50 i am running an xlite and configured a user say 200 with proxy as 172.16.7.63.I
 modified the sip.conf file with user [200].When i run xlite in asterix cli i am able to see the mesage Registered 200 from 172.16.7.50.But when i give the command sip show registry nothing is being displayed?How to proceed on this.


2.From another system i want to run one more xlite client say 172.16.7.62.Again the asterisk cli shows Registered 300 user from 172.16.7.50.When i make a call from 200 to 300 it says 404 not found.Is my approach correct.I
 am completely new to this technology and please guide me how to proceed on this.

Thanks  Regards,
Saritha.
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Re: [asterisk-users] stress a server with a tool

2006-09-20 Thread BJ Weschke


On 9/20/06, nik600 [EMAIL PROTECTED] wrote:

hi

is there any software usable to simulate work on an asterisk server?

I'm interested in it to evaluate the level of currently calls that a
server can support


For SIP, see http://sipp.sourceforge.net/

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[asterisk-users] Register doubt

2006-09-20 Thread god

Sir,  I installed asterix in some system say with ip 172.16.7.63.From someother windows system say 
172.16.7.50 i am running an xlite and configured auser say 200 with proxy as 172.16.7.63.I modified the sip.conf file withuser [200].When i run xlite in asterix cli i am able to see the mesageRegistered 200 from 
172.16.7.50.But when i give the command sip showregistry nothing is being displayed?How to proceed on this.2.From another system i want to run one more xlite client say172.16.7.62.Again the asterisk cli shows Registered 300 user from
172.16.7.50.When i make a call from 200 to 300 it says 404 not found.Is myapproach correct.I am completely new to this technology and please guide mehow to proceed on this.Thanks  Regards,Saritha.

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Re: Re: [asterisk-users] Uninstalling Trixbox

I Installed it using the rpm that is available for download on trixbox website along with the ISO image. Well actually its on sourceforge.net. anyways, unlike the ISO image, rpm package only installs trixbox and asterisk not the operating system. its for linux.

On 9/20/06, Mike Dent [EMAIL PROTECTED] wrote:
When you installed Trixbox did you not boot from the Trixbox install CD?This installs CentOS and Trixbox.
I'm curious how you installed Trixbox?MikeOn 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote: no, im not using FreePBX, actually freepbx is a part of trixbox as is
sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE. By saying that 'It has taken control of my system', i meant asterisk. Now i dont want any web based interface to asterisk. i only want asterisk on my
system. So plz help me uninstall trixbox.And Sharon, thanx for the tip, but what about the rest of the scripts trixbox has installed on my system. for example i dont want to start asterisk on system startup, but trixbox does that. so anymore help will be
helpfull :) On 9/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote:
Hi all, trixbox has taken control of my asterisk system, i dont like that. i just installed trixbox for rersearch purpose now i want to uninstall it and do
some research on asterisk. So plz tell me how to uninstall trixbox. will it uninstall asterisk also? Trixbox has not taken over your system. It is your system. It is not
just the web interface. Trixbox is a customized CentOS distribution. It uses a number of its own packages in its own yum source (in addition to standard CentOS packages), and has a number of non-default settings.
e.g: selinux disabled. Or you may be confusing it with FreePBX? -- Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755iax:[EMAIL PROTECTED] +972-50-7952406jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by
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http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___
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Re: [asterisk-users] Uninstalling Trixbox

2006-09-20 Thread Avi Miller


Mike Dent wrote:

I'm curious how you installed Trixbox?


There is a tar.gz version of Trixbox that can be installed over an 
existing RHEL4 or CentOS installation.


However, removing Trixbox is very difficult. You are better off 
reinstalling RHEL4 and then installating Asterisk from scratch.


cYa,
Avi

--
National Manager - Special Projects

 Sydney / Melbourne / Canberra / Hobart / London /
   2/340 Gore Street  T: 1 3000 SQUIZ (77849)
   Fitzroy, VIC   T: +61 (0) 3 9235 5400
   3065   F: +61 (0) 3 9235 5444
  W: http://www.squiz.net/

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Re: [asterisk-users] Uninstalling Trixbox

2006-09-20 Thread bails


rpm -e packagename ?

Rizwan Hisham wrote:

I Installed it using the rpm that is available for download on trixbox 
website along with the ISO image. Well actually its on sourceforge.net 
http://sourceforge.net. anyways, unlike the ISO image, rpm package 
only installs trixbox and asterisk not the operating system. its for 
linux.


On 9/20/06, *Mike Dent* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
wrote:


When you installed Trixbox did you not boot from the Trixbox
install CD?
This installs CentOS and Trixbox.
I'm curious how you installed Trixbox?

Mike


On 9/20/06, Rizwan Hisham [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
 no, im not using FreePBX, actually freepbx is a part of trixbox
as is
 sugarCRM, FOP etc. and i also dont know about CentOS, im using
RHL EE. By
 saying that 'It has taken control of my system', i meant
asterisk. Now i
 dont want any web based interface to asterisk. i only want
asterisk on my
 system. So plz help me uninstall trixbox.

  And Sharon, thanx for the tip, but what about the rest of the
scripts
 trixbox has installed on my system. for example i dont want to start
 asterisk on system startup, but trixbox does that. so anymore
help will be
 helpfull :)


 On 9/20/06, Tzafrir Cohen [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
  On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote:
   Hi all,
   trixbox has taken control of my asterisk system, i dont like
that. i
 just
   installed trixbox for rersearch purpose now i want to
uninstall it and
 do
   some research on asterisk. So plz tell me how to uninstall
trixbox. will
 it
   uninstall asterisk also?
 
  Trixbox has not taken over your system. It is your system. It
is not
  just the web interface. Trixbox is a customized CentOS
distribution. It
  uses a number of its own packages in its own yum source (in
addition to
  standard CentOS packages), and has a number of non-default
settings.
  e.g: selinux disabled.
 
  Or you may be confusing it with FreePBX?
 
  --
  Tzafrir Cohen sip:[EMAIL PROTECTED]
mailto:sip:[EMAIL PROTECTED]
  icq#16849755  iax:[EMAIL PROTECTED]
mailto:iax:[EMAIL PROTECTED]
  +972-50-7952406  jabber:[EMAIL PROTECTED]
mailto:jabber:[EMAIL PROTECTED]
  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http://www.xorcom.com

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[asterisk-users] University dumps CISCO VoIP for Asterisk

2006-09-20 Thread Ferguson, Michael




G'Day 
List,

Interesting article. 
Enjoy

http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5

Mike
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Re: [asterisk-users] Uninstalling Trixbox

sorry i made a mistake telling you that i installed it using rpm package. actually there is no rpm package for trixbox to download. you have to install it using .tar.gz package using ur existing linux OS. so sorry about that.

On 9/20/06, bails [EMAIL PROTECTED] wrote:
rpm -e packagename ?Rizwan Hisham wrote: I Installed it using the rpm that is available for download on trixbox
website along with the ISO image. Well actually its on sourceforge.net http://sourceforge.net. anyways, unlike the ISO image, rpm package
only installs trixbox and asterisk not the operating system. its for linux. On 9/20/06, *Mike Dent* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: When you installed Trixbox did you not boot from the Trixbox install CD? This installs CentOS and Trixbox. I'm curious how you installed Trixbox?
Mike On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote: no, im not using FreePBX, actually freepbx is a part of trixbox as is sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE. By
saying that 'It has taken control of my system', i meant asterisk. Now i dont want any web based interface to asterisk. i only want asterisk on my system. So plz help me uninstall trixbox.
And Sharon, thanx for the tip, but what about the rest of the scripts trixbox has installed on my system. for example i dont want to start asterisk on system startup, but trixbox does that. so anymore
help will be helpfull :)On 9/20/06, Tzafrir Cohen [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote:Hi all,trixbox has taken control of my asterisk system, i dont like
that. i justinstalled trixbox for rersearch purpose now i want to uninstall it and dosome research on asterisk. So plz tell me how to uninstall
trixbox. will ituninstall asterisk also? Trixbox has not taken over your system. It is your system. It is not
just the web interface. Trixbox is a customized CentOS distribution. It uses a number of its own packages in its own yum source (in addition to standard CentOS packages), and has a number of non-default
settings. e.g: selinux disabled. Or you may be confusing it with FreePBX? -- Tzafrir Cohen
sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] icq#16849755
iax:[EMAIL PROTECTED] mailto:iax:[EMAIL PROTECTED] +972-50-7952406
jabber:[EMAIL PROTECTED] mailto:jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.xorcom.com ___
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[asterisk-users] enumlookup - deprecated working - but appreciated one duznt :-(

2006-09-20 Thread Benjamin Jacob



Hello ppl,

I had appdata set to use the function ENUMLOOKUP. But it gets me nothing.
| id|   context | exten   |  priority | app   | 
   appdata
  
==
48 |   pbx1| _011.   | 1| Set   
| enumresult=${ENUMLOOKUP(+13015611020,sip,c,enum.info)}
| 49   |   pbx1| _011.   | 2| SayDigits | 
   ${enumresult} 


But, using the application, EnumLookup, I do get back the results.
| 48   |   pbx1| _011.   | 1| EnumLookup  | 
+13015611020
| 49   |   pbx1| _011.   | 2| Dial  
| ${ENUM} 

Another interesting observation, in my enum.conf, I've set only search 
= enum.info .
In the tcpdump, I see EnumLookup, the deprecated one looking for the 
correct enum.info, but, with the function ENUMLOOKUP, I see enum.arpa 
being pinged!!???


Any ideas where I am going wrong?

My enum.info pasted :
===
;
; ENUM Configuration for resolving phone numbers over DNS
;
; Sample config for Asterisk
; This file is reloaded at reload enum in the CLI
;
[general]
;
; The search list for domains may be customized.  Domains are searched
; in the order they are listed here.
;
;search = e164.arpa
;
; If you'd like to use the E.164.org public ENUM registery in addition
; to the official e164.arpa one, uncomment the following line
;
;search = e164.org
search = e164.info
;
; As there are more H323 drivers available you have to select to which
; drive a H323 URI will map. Default is H323.
;
h323driver = H323

==
I got the enum.info info, from the site, http://nona.net/features/enum/ .

cheerz
Ben.


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[asterisk-users] IAX2 register refuse but Dial cmd works!

2006-09-20 Thread Ma Zhiyong

Hi, I just set two asterisk connect with iax2 trunk.

B server 
[user1]
type=user
trunk=yes
context=from-trunk
username=user1
auth=plaintext
secret=passwd
notransfer=yes

A server 
register = user1:[EMAIL PROTECTED]

I notice  on A's CLI, it  shows Registration of 'user1' rejected: 
'Registration Refused' from: 'x.x.x.x'. I also use iax2 show registry it say 
Unregistered.

But when I use dial cmd: Dial(iax2/user1:[EMAIL PROTECTED]/${exten},30,), I can 
call the extension normally.

What's wrong? Why can't I register to B server?
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[asterisk-users] Forwarding the Ring Group and Calls coming in to Queues

2006-09-20 Thread Zeeshan Zakaria

My client needs an option to forward the incoming calls to the ring group and to the queues to his other number when he closes his office. By default, at 9 PM the incoming calls are forwarded automatically. But he wants something so that if he closes earlier, he can forward the incoming calls manually, or if nobody answers any of the ring group phones in regular hours, after 60 sec instead of activating voicemail, call should be forwarded to his other office number.


First, he has option 0 which activated the ring group of 4 extensions. How can I do so that he can manually forward this ring group to his other office number, so that when caller presses 0, call doesn't go to the ring group and is forwarded instead.


Second, on pressing other options, like 2, or 3, queues are activated. How can I do that he can manually forward these incoming calls to the other number.

All this I can do for individual extensions, but don't know how to do it for the ring group or for the queues. Please guide me on how to do this.
-- Zeeshan A Zakaria 
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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread burke

   I would like to know how you got Asterisk to function with 2500 SIP
 registrations.  Did you have qualify enabled?

 Yes, qualify was enabled, using the standard length of qualification
 period between checks. Very few accounts had custom qualify settings.

 What about the 500 simultaneous calls?  How many SQL hits were you
 doing (all said and done).  Any performance logs from the SQL server?

 I can't believe you got all this running on one box!

 You have to remember, 500 simultaneous calls is not the same as
 something like 20 calls per second. some of those calls may have been
 quite long, and once the call's been placed, there's no database work
 being done until the call ends.

 I wish I had statistics from that setup, but I don't, we spent so much
 time implementing new features and chasing down problems caused by using
 a pre-RTA version of Asterisk with a patched in RTA setup.



 --
 S McGowan
 VoIP Consultant
 [EMAIL PROTECTED]


S McGowan,

I don't know if you missed my question (from the slew of questions you've
received and answered), but I was wondering about transcoding and PSTN
channels. What kind of codecs were used and was there any transcoding
happening? Was this box only responsible for VoIP-to-VoIP calls or was
there also PSTN trunks as well? Again, I'm amazed by this example since it
seems to be way over what anyone else normally reports as usable.

Thanks again,
Ryan
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[asterisk-users] Incoming calls, identify

2006-09-20 Thread joea, j4computers

Just delving into asterisk, using trixbox 1.2 and a TDM400p.  The card will 
have two FXO and two FXS modules.

Two incoming analog lines, which need to be treated as distinct entities.  
Meaning, for example, line 1= company1, line2=company2, or line 1= home line, 
line2=business line.  In my limited setup and testing, did not see (obviously) 
how to do this.  I would think this is fundamental and only my new-ish-ness is 
in the way.

Also, as the card does not have any docs with it, what is the power connector 
for?  Is this necessary to allow ringer power to be supplied to analog phones?

joe

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RE: [asterisk-users] Digium GUI?

2006-09-20 Thread Douglas Garstang

So, is this GUI you speak of so often able to cater to CARRIERs rather than 
ENTERPRISEs?

-Original Message- 
From: shadowym [mailto:[EMAIL PROTECTED] 
Sent: Tue 9/19/2006 10:47 PM 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Cc: 
Subject: RE: [asterisk-users] Digium GUI?

There is the underlying framework for developers to do their own thing 
but Digium has also made their own GUI.  It's a GUI, a REAL GUI!  it's in the 
FAQ's and press releases.  In other words it's public knowledge.  A GUI like 
FreePBX.  In other words, a GUI!  Did I mention it's a GUI!  Not just a 
framework for a GUI but also an actual GUI.

Did I mention that it is a GUI!  A REAL GUI.  It was on display at VON! 
 A GUI as in point the mouse and click kind of thing.  I believe that is called 
a GUI.  It's graphical, and the user interfaces with it.  They call that a GUI.

  _  

From: mitcheloc [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, September 19, 2006 4:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digium GUI?

No it's not, it's supposed to just be a framework for developers and 
resellers to create GUIs that can go on the appliance.

On 9/19/06, shadowym [EMAIL PROTECTED] wrote: 

I am talking about the GUI that was announced as part of the 
new Asterisk
Appliance.

Sounds like it is going to be a full featured GUI like FreePBX. 

-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Monday, September 18, 2006 8:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: Re: [asterisk-users] Digium GUI?

 So the press announcement said that the new Digium GUI will be
 available in
 v1.4 sometime in Oct.  Is the GUI already there in Trunk or 
is there
 some other branch of development that the general public 
cannot access?

Do you mean this?

http://svn.digium.com/view/asterisk/trunk/static-http/ 

On 9/18/06, Don Fanning [EMAIL PROTECTED] wrote:
 You mean the menuselect ncurses screen?  If yes, then yes... 
it's a
 gui. :)

 -Original Message- 
 From: shadowym [mailto:[EMAIL PROTECTED]
 Sent: Monday, September 18, 2006 4:43 PM
 To: asterisk-users@lists.digium.com 
 Subject: [asterisk-users] Digium GUI?

 So the press announcement said that the new Digium GUI will be
 available in
 v1.4 sometime in Oct.  Is the GUI already there in Trunk or 
is there 
 some other branch of development that the general public 
cannot access?

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-- 

Mitchel Constantin
Snap - A desktop user interface for Asterisk
www.snapanumber.com 

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Re: [asterisk-users] University dumps CISCO VoIP for Asterisk

2006-09-20 Thread joea, j4computers

Ferguson, Michael[EMAIL PROTECTED] Wrote on: 9/20/2006 8:03 AM:
 G'Day List,
  
 Interesting article. Enjoy
  
 http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5 
  
 Mike

The text states that asterisk cannot do secretarial functions, meaning one 
person, or admin, cannot answer multiple lines.  

This relates a bit to my recent post, asking about servicing multiple lines.  

Implication is that asterisk can do that, but I am now concerned that there is 
no uber function that can allow a single person answer any line, for reasons 
of convenience or design.  Problem is, this was understood, rightly or wrongly, 
to exist, in preliminary inquiries (not here) and is a part of a potential 
clients desire.

Can someone enlighten me?

joe
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[asterisk-users] Realtime madness

2006-09-20 Thread Scott Pinhorne


Hi All

I have 2 sip users setup in the database for realtime and they also have 
their extension setup in the database.


When I register user 1 fine and can make and recieve calls.
As soon as i register user2 user1 is then unable to make any calls??

If i put the config fr both users in the flat config files and register 
them both it works fine, its only when they are running in realtime from 
database.


anyone knwo whats going? a comand line output doesnt shown anything for 
user1 when user2 is registered.


thanks
scott
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Re: [asterisk-users] Channel kept busy when creating ssh tunnel via AGI

2006-09-20 Thread Michiel van Baak

On 06:25, Wed 20 Sep 06, BJ Weschke wrote:
 If you don't want such behavior, you might want to take a look at
 monitoring a specific channel event in the Asterisk manager and then
 starting off your script upon the receipt of such an event through the
 manager.

Or if you want to go dirty:
run an agi that looks lik this:

#!/bin/sh
/path/to/my/actual/script
exit 0

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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RE: [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8

2006-09-20 Thread Jeronimo Romero

This is pound key linux from rpath. I don't see a source directory. That
is why I think I must be missing something. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, September 19, 2006 10:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH distorted on Pound Key Linux on
asterisk1.2.8

Remove mpg123.  In the Asterisk source directory type make mpg123  I

believe that make install is required to install it.

Jeronimo Romero wrote:
 Running Asterisk 1.2.8 on Pound Key linux which I downloaded from
Digium
 site. 

 Uname output: 

 Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov 23 11:31:48 EST
2005
 i686 athlon i386 GNU/Linux

 It didn't come with mpg123 so I downloaded it from the internet.  MOH
 works, but it is terribly  loud and mistorted. Tried running under
 quitemp3 profile but it didn't help. 

 I feel like there is something I may be missing here. Any ideas???
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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr

 S McGowan,
 
 I don't know if you missed my question (from the slew of questions you've
 received and answered), but I was wondering about transcoding and PSTN
 channels. What kind of codecs were used and was there any transcoding
 happening? Was this box only responsible for VoIP-to-VoIP calls or was
 there also PSTN trunks as well? Again, I'm amazed by this example since it
 seems to be way over what anyone else normally reports as usable.
 
 Thanks again,
 Ryan


Ryan, I answered, but for some reason this pop account tends to be
strange... Anyway, we were not doing any transcoding and our PSTN
connectivity was handled via a Tier 1 ISP that does SIP only PSTN
connectivity solutions with G.711u. So, basically as far as Asterisk was
concerned, there was SIP and RDP, that's all.


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Re: [asterisk-users] University dumps CISCO VoIP for Asterisk

joea, j4computers wrote:

Ferguson, Michael[EMAIL PROTECTED] Wrote on: 9/20/2006 8:03 AM:

G'Day List,

Interesting article. Enjoy

http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5

Mike

The text states that asterisk cannot do secretarial functions, meaning one person, or admin, cannot answer multiple lines.

This relates a bit to my recent post, asking about servicing multiple lines.

Implication is that asterisk can do that, but I am now concerned that there is no
uber function that can allow a single person answer any line, for reasons of
convenience or design. Problem is, this was understood, rightly or wrongly, to exist, in
preliminary inquiries (not here) and is a part of a potential clients desire.

Can someone enlighten me?

The problem with a BLF (Busy Lamp Field) is that it's hard to find a box
with 6,000 buttons on it, as would be required by above university.

Asterisk has several methods of picking up remote lines. Group Call
Pickup, Directed Call Pickup, and the standard way Asterisk rings
multiple extensions at the same time via in the Dial() Line, and BLF

If you want the traditional Key System style of BLF, then you need a
phone that supports it. The Polycom 601 Sidecar supports it in a
limited way, and I've heard that SNOM supports it as well.

What SPECIFICALLY are you trying to do that you are unable to do?

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[asterisk-users] Sip configuration using mysql

2006-09-20 Thread Arkaitz


Hi,
I'm trying to use mysql for sip users management and i'm a bit stuck
with a problem.
I use asterisk-1.2.12.1 and res_config_mysql from asterisk-addons-1.2.4.
The fact is that i've put a row in the mysql sip table for my linksys
phone and i can make calls and receive calls with it, but it doesn't
appear in sip show peers, and asterisk is unable to find files when
I use that phone configured from mysql.

Sep 20 15:11:24 WARNING[8347]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to slin
Sep 20 15:11:24 WARNING[8347]: app_festival.c:187
send_waveform_to_channel: Unable to set write format to signed linear
Sep 20 15:22:27 WARNING[8370]: channel.c:2752
ast_channel_make_compatible: No path to translate from
SIP/saladino-081aa1c8(4) to SIP/linksys-a6f017a0(256)
Sep 20 15:22:48 WARNING[8376]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to gsm
Sep 20 15:22:48 WARNING[8376]: file.c:824 ast_streamfile: Unable to
open vm-intro (format g729): No such file or directory

When i configure it from sip.conf file it works perfect (i comment the
entry when i want to use the mysql conf).
[linksys]
callerid=linksys
type=friend
user=linksys
secret=linksys
context=saladino
host=dynamic

The mysql part:
mysql desc sip;
++--+--+-+-++
| Field  | Type | Null | Key | Default
| Extra  |
++--+--+-+-++
| id | int(11)  | NO   | PRI | NULL
| auto_increment |
| name   | varchar(80)  | NO   | UNI |
||
| accountcode| varchar(20)  | YES  | | NULL
||
| amaflags   | varchar(13)  | YES  | | NULL
||
| callgroup  | varchar(10)  | YES  | | NULL
||
| callerid   | varchar(80)  | YES  | | NULL
||
| canreinvite| char(3)  | YES  | | yes
||
| context| varchar(80)  | YES  | | NULL
||
| defaultip  | varchar(15)  | YES  | | NULL
||
| dtmfmode   | varchar(7)   | YES  | | NULL
||
| fromuser   | varchar(80)  | YES  | | NULL
||
| fromdomain | varchar(80)  | YES  | | NULL
||
| fullcontact| varchar(80)  | YES  | | NULL
||
| host   | varchar(31)  | NO   | |
||
| insecure   | varchar(4)   | YES  | | NULL
||
| language   | char(2)  | YES  | | NULL
||
| mailbox| varchar(50)  | YES  | | NULL
||
| md5secret  | varchar(80)  | YES  | | NULL
||
| nat| varchar(5)   | NO   | | no
||
| deny   | varchar(95)  | YES  | | NULL
||
| permit | varchar(95)  | YES  | | NULL
||
| mask   | varchar(95)  | YES  | | NULL
||
| pickupgroup| varchar(10)  | YES  | | NULL
||
| port   | varchar(5)   | NO   | |
||
| qualify| char(3)  | YES  | | NULL
||
| restrictcid| char(1)  | YES  | | NULL
||
| rtptimeout | char(3)  | YES  | | NULL
||
| rtpholdtimeout | char(3)  | YES  | | NULL
||
| secret | varchar(80)  | YES  | | NULL
||
| type   | varchar(6)   | NO   | | friend
||
| username   | varchar(80)  | NO   | |
||
| disallow   | varchar(100) | YES  | | all
||
| allow  | varchar(100) | YES  | | g729;ilbc;gsm;ulaw;alaw
||
| musiconhold| varchar(100) | YES  | | NULL
||
| regseconds | int(11)  | NO   | | 0
||
| ipaddr | varchar(15)  | NO   | |
||
| regexten   | varchar(80)  | NO   | |
||
| cancallforward | char(3)  | YES  | | yes
||
| setvar | varchar(100) | NO   | |
||
++--+--+-+-++

Phone row:
id=2
name=linksys
canreinvite=yes
context=saladino
dtmfmode=rfc2833
host=dynamic
nat=yes
secret=linksys
type=peer
username=linksys
disallow=all
allow=g729;ilbc;gsm;ulaw;alaw

Other fields are NULL.

Any hint?
Thanks for your time.

--
Arkaitz
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Re: [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8

I can't help you with distro specific stuff.  You need MPG123 1.59r  If 
you do not have that version then you will experience these issues.  OR 
you could use the Native MOH features of Asterisk.


Jeronimo Romero wrote:

This is pound key linux from rpath. I don't see a source directory. That
is why I think I must be missing something. 




-Original Message-
From: [EMAIL PROTECTED]

[mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling

Sent: Tuesday, September 19, 2006 10:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH distorted on Pound Key Linux on
asterisk1.2.8



Remove mpg123.  In the Asterisk source directory type make mpg123  I



believe that make install is required to install it.



Jeronimo Romero wrote:

Running Asterisk 1.2.8 on Pound Key linux which I downloaded from

Digium
site. 

Uname output: 


Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov 23 11:31:48 EST

2005

i686 athlon i386 GNU/Linux

 

 


It didn't come with mpg123 so I downloaded it from the internet.  MOH
works, but it is terribly  loud and mistorted. Tried running under
quitemp3 profile but it didn't help. 

 


I feel like there is something I may be missing here. Any ideas???

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Re: [asterisk-users] Channel kept busy when creating ssh tunnel via

2006-09-20 Thread Doug Lytle


Michiel van Baak wrote:

On 06:25, Wed 20 Sep 06, BJ Weschke wrote:
  
#!/bin/sh

/path/to/my/actual/script
exit 0
  


If you were to do that, then you might as well use System()

Doug

--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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[asterisk-users] tx_fax over sip to TDM card

2006-09-20 Thread Jerry Geis


Is it not possible to run tx_fax over a SIP connection
to another box that dials using a TDM card?

My is seg faulting.

I can make phone calls over that same SIP connection
so everything is working there just not tx_fax?

Any idea? or is this not supported.

Jerry

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RE: [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8

2006-09-20 Thread Jeronimo Romero

Yes. It this is the opensource poundkey from rpath. I just installed madplay 
instead of dealing with mpg123. Works like a charm.  Is there any downside to 
madplay that that I should know about??

Here my musiconhold.conf file:


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of brandon kruz
 Sent: Tuesday, September 19, 2006 10:46 PM
 To: asterisk-users@lists.digium.com
 Subject: RE: [asterisk-users] MOH distorted on Pound Key Linux on
 asterisk1.2.8
 
 is this open source poundkey?
 
 and can i see your moh conf??
 
 im guessing its open source pk since you mentioned the asterisk 1.2.8
 part.
 
 and also is it the default MOH or your own cooked up version??
 also i recommend, if not necessary the EXACT mpg version described in the
 conf
 (321 0.9?) something similar, please try this, but first of all before we
 go
 that extreme.
 
 lets see your conf's and if you made your own music, or default MOH
 From: Jeronimo Romero [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial
 Discussionasterisk-users@lists.digium.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] MOH distorted on Pound Key Linux on asterisk
 1.2.8
 Date: Tue, 19 Sep 2006 20:34:20 -0400
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 Running Asterisk 1.2.8 on Pound Key linux which I downloaded from Digium
 site.
 
 Uname output:
 
 Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov 23 11:31:48 EST 2005
 i686 athlon i386 GNU/Linux
 
 
 
 
 
 It didn't come with mpg123 so I downloaded it from the internet.  MOH
 works, but it is terribly  loud and mistorted. Tried running under
 quitemp3 profile but it didn't help.
 
 
 
 I feel like there is something I may be missing here. Any ideas???
 
 Thanks in advance.
 
 
 
 Jeronimo.
 
 
 
 
 
 ==
 
 Jeronimo Romero
 
 EUS Networks
 
 Email: [EMAIL PROTECTED]
 
 Cell: 917-332-7238
 
 Office: 212-624-5943
 
 Web: www.euscorp.com
 
 ==
 
 
 
 
 
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[asterisk-users] Unexpected delay

2006-09-20 Thread flavio


Hi to all.

I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and
I've some problem with outgoing calls: there is a big delay for
bidirectional audio flow.

Here is mean part of an asterisk trace releted to outgoing calls.
(canreinvite=no for both peers).
Until SIP 180 ringing signaling is correct...bold highlight time for NOTICE

_ _ _ _




Sep 18 16:01:43 [1;33;40mNOTICE[0;37;40m[23098]:
[1;37;40mchan_sip.c[0;37;40m:[1;37;40m9854[0;37;40m
[1;37;40mhandle_response_register[0;37;40m: Outbound Registration:
Expiry for 10.28.52.74 is 3599 sec (Scheduling reregistration in 3584
s)

[1;30;40m -- [0;37;40mSIP/outgoing-08197388 is ringing

Transmitting (no NAT) to 10.28.52.244:5060:
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP
10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244
From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482
To: sip:[EMAIL PROTECTED];user=phone;tag=as2ea0ddd1
Call-ID: [EMAIL PROTECTED]
CSeq: 829 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
---

[1;30;40m -- [0;37;40mSIP/outgoing-08197388 is making progress passing
it to SIP/bt102-08190d90
Sep 18 16:02:37 [1;33;40mNOTICE[0;37;40m[23098]:
[1;37;40mchan_sip.c[0;37;40m:[1;37;40m11613[0;37;40m
[1;37;40msip_poke_noanswer[0;37;40m: Peer 'outgoing' is now
UNREACHABLE! Last qualify: 4


-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

[1;30;40m -- [0;37;40mSIP/outgoing-08197388 answered SIP/bt102-08190d90

We're at 10.28.52.246 port 16274

Adding codec 0x4 (ulaw) to SDP

Adding codec 0x8 (alaw) to SDP

Reliably Transmitting (no NAT) to 10.28.52.244:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244
From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482
To: sip:[EMAIL PROTECTED];user=phone;tag=as2ea0ddd1
Call-ID: [EMAIL PROTECTED]
CSeq: 829 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 184
v=0
o=root 23109 23110 IN IP4 10.28.52.246
s=session
c=IN IP4 10.28.52.246
t=0 0
m=audio 16274 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---

[1;30;40m -- [0;37;40mAttempting native bridge of SIP/bt102-08190d90
and SIP/outgoing-08197388
Sep 18 16:03:25 [1;33;40mNOTICE[0;37;40m[23098]:
[1;37;40mchan_sip.c[0;37;40m:[1;37;40m9882[0;37;40m
[1;37;40mhandle_response_peerpoke[0;37;40m: Peer 'outgoing' is now
REACHABLE! (6ms / 2000ms)

-- SIP read from 10.28.52.244:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bK30db550457acdb99
From: sip:[EMAIL PROTECTED];user=phone;tag=a82e9be13c882482
To: sip:06720228.52.246;user=phone;tag=as2ea0ddd1
Contact: sip:[EMAIL PROTECTED];user=phone
Call-ID: [EMAIL PROTECTED]
CSeq: 829 ACK
User-Agent: Grandstream BT110 1.0.8.12
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

_ _ _ _


From trace it points out that time gap from 180 Ringing and follow 200

Ok is about 1 minute.. and so from 200 OK and ACK
Any suggestions?

Moreover..when I attempt to make an outgoing call with option
canreinvite=yes, Asterisk notifies the follow message?

Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked:
Avoided initial deadlock for '0x819b240', 10 retries!

Can anyone tell me what it does mean and how to fix it?


Thanks 4 all


--

* (o ing. Patria Flavio
* //\  phone 0823451358
* V_/_  mobile 3407873357
*

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Re: [asterisk-users] Realtime madness

2006-09-20 Thread Michel Vaillancourt

Scott Pinhorne wrote:
 Hi All
 
 I have 2 sip users setup in the database for realtime and they also have
 their extension setup in the database.
 
 When I register user 1 fine and can make and recieve calls.
 As soon as i register user2 user1 is then unable to make any calls??
 
 If i put the config fr both users in the flat config files and register
 them both it works fine, its only when they are running in realtime from
 database.
 
 anyone knwo whats going? a comand line output doesnt shown anything for
 user1 when user2 is registered.
 
 thanks
 scott

Not really sure what is going on here.  We use ARI for everything. 
There 40 phones defined in our office set up, for example, and  call routing 
never hitches up.  Can you post a sanitized SELECT * of your SIP user table?

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+1 514 395 1106 ext 117
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Re: [asterisk-users] Sip configuration using mysql

2006-09-20 Thread Michel Vaillancourt

Arkaitz wrote:
 Hi,
 I'm trying to use mysql for sip users management and i'm a bit stuck
 with a problem.
 I use asterisk-1.2.12.1 and res_config_mysql from asterisk-addons-1.2.4.
 The fact is that i've put a row in the mysql sip table for my linksys
 phone and i can make calls and receive calls with it, but it doesn't
 appear in sip show peers, and asterisk is unable to find files when
 I use that phone configured from mysql.
 
Try:
/etc/asterisk/sip.conf
[general]
rtcachefriends=yes


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RE: [asterisk-users] Format_MP3, Streaming, File Formats, MOH

2006-09-20 Thread Asterisk [Submusic]










Hi,



This config is working for me:



_



musiconhold.conf



[shoutcast]

mode=custom

application=/usr/local/bin/mpg123 -s --mono -y -f
8192 -r 8000 http://stream128.submusic.ch:8004/



; The '/' in the stream URL is important !



_



extensions.conf



exten = 17,1,Answer

exten = 17,2,MusicOnHold(shoutcast)



_





Regards





Frederic













De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Robert Chadwell
Envoy: mardi, 19. septembre
2006 14:47
:
asterisk-users@lists.digium.com
Objet: [asterisk-users]
Format_MP3, Streaming, File Formats, MOH





Format_MP3 appears to play MOH files starting at the
beginning of each file, using the .wav file format, making for some repetitive
hold music unless you alter the file itself to begin somewhere in the middle.



Solution: One stream that all users connect to
 giving dynamic hold music (tried and tested in A1.0x using mpg123 with
some success, and Icecast or Slimserver or Shoutcast)



Format_MP3 doesnt seem to stream, and the wiki
is wrong about streamplayer being used to play streams, as it is only used to
play raw TCP streams. 



There are many
questions in forums on the web with no answers about how to solve this dilemma,
How do you get users connected to a constantly-changing stream of music instead
of streams starting from the beginning (regardless of whether Linux counts them
as one stream or not where the processor is concerned)?



Hopefully, at the end of this thread, I will have
enough information to go back to these web-forums and post the answer. To get
it started  here is what I have tried that hasnt worked. In most
all cases the response is Music on hold started, immediately
followed by Music on hold stopped with no sound in any case.



;[classes]

;mode=custom

;application=/usr/bin/streamplayer 194.158.114.67
8000

;format=ulaw

--- Straight From The Music On Hold Wiki



;default =
quietmp3:/var/lib/asterisk/mohmp3-dummy
-@,http://www.shoutcast.com/sbin/tunein-station.pls?id=7733filename=playlist.pls

--- From the Nerd Vittles Tutorial with the
-@ added because mpg123 seemed to ask for it since the file was a .pls



;default = mp3:http://127.0.0.1:9000/stream.mp3

-- From a forum of someone using mpg123 to
stream SlimServer (installed mpg123 v0.60 with no success here)



[default]

mode=files

directory=
/var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3

-- Tried a 1.2 format



;default =
quietmp3:/var/lib/asterisk/mohmp3-dummy,http://193.251.154.243:8000/

-- Thought maybe it was SlimServer  so
tried to stream the top Shoutcast station



;default =
quietmp3:/var/lib/asterisk/mohmp3-dummy,http://127.0.0.1:9000/stream.mp3

-- Tried to stream Slimserver using the old
format





Thank you in
advance  I have been at this for a week now. How did you make it work in
Asterisk 1.2x?



Rob








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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Kristian Kielhofner


Rushowr wrote:

S McGowan,

I don't know if you missed my question (from the slew of questions you've
received and answered), but I was wondering about transcoding and PSTN
channels. What kind of codecs were used and was there any transcoding
happening? Was this box only responsible for VoIP-to-VoIP calls or was
there also PSTN trunks as well? Again, I'm amazed by this example since it
seems to be way over what anyone else normally reports as usable.

Thanks again,
Ryan




Ryan, I answered, but for some reason this pop account tends to be
strange... Anyway, we were not doing any transcoding and our PSTN
connectivity was handled via a Tier 1 ISP that does SIP only PSTN
connectivity solutions with G.711u. So, basically as far as Asterisk was
concerned, there was SIP and RDP, that's all.



	So there was 2500 SIP registrations with qualify, 500 active calls with 
SIP and RTP, realtime, and CDR logging via MySQL (all on the same box)?


What source changes did you make?  What OS tweaks?

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Asterisk capabilities, was [asterisk-users] University dumps CISCO VoIP for Asterisk

2006-09-20 Thread joea, j4computers

 . . 
 
 What SPECIFICALLY are you trying to do that you are unable to do?
 

No specifics, at this time, too early in evaluation.  

I get the point, I think, about thousands of buttons.  

My concerns are the ability to answer on multiple lines, and have various 
options,upon no pickup, including have selected (or all) unanswered calls ring 
thru to, or be picked up by, an admin or catch all operator.

I was assured this was cake for asterisk, but was concerned by the article.

joea
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Re: [asterisk-users] Tracking the source of a disconnect? - SOLVED

2006-09-20 Thread Jamin W. Collins


Doug Lytle wrote:

Jamin W. Collins wrote:

 callprogress = yes

The only thing I'm iffy about is the above entry.

Maybe it's mistaking the progress as disconnect?


That does appear to have been the issue.  We haven't had a new 
occurrence of the random disconnects since disabling callprogress.


Thank you.

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Kristian Kielhofner


[EMAIL PROTECTED] wrote:

Again, I'm amazed by this example since it

seems to be way over what anyone else normally reports as usable.


Exactly!

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Re: [asterisk-users] Sip configuration using mysql

2006-09-20 Thread Arkaitz


Hi,
Thanks, now i see the phone in show sip peers, I've been reading
about rtcachefriends and now i understand what was the problem.
But the other problem is still here :(. It seems that asterisk is
unable to find any file in the system, not gsm file nor codec...
nothing.  It's strange since i provide the same options in sip.conf
than in mysql row, but still it fails. i don't understand why.
Thanks for your time


On 9/20/06, Michel Vaillancourt [EMAIL PROTECTED] wrote:

Arkaitz wrote:
 Hi,
 I'm trying to use mysql for sip users management and i'm a bit stuck
 with a problem.
 I use asterisk-1.2.12.1 and res_config_mysql from asterisk-addons-1.2.4.
 The fact is that i've put a row in the mysql sip table for my linksys
 phone and i can make calls and receive calls with it, but it doesn't
 appear in sip show peers, and asterisk is unable to find files when
 I use that phone configured from mysql.

Try:
/etc/asterisk/sip.conf
[general]
rtcachefriends=yes


--
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Senior Telephony Engineer
Neoxo Inc  (www.neoxo.com)
+1 514 395 1106 ext 117
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--
Arkaitz
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Re: [asterisk-users] Format_MP3, Streaming, File Formats, MOH

2006-09-20 Thread Raphaël Jacquot

Asterisk [Submusic] wrote:

 musiconhold.conf
 [shoutcast]
 mode=custom
 application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000
 http://stream128.submusic.ch:8004/
 ; The  '/' in the stream URL is important !

I tried this.
however it doesn't work. apparently, asterisk doesn't read from the
mpg123 when no one is listening to MOH, and stuff appear to be stacking
inside a pipe of some sort.
when the next caller gets the MOH, he gets the music from 5 minutes ago
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Re: Asterisk capabilities, was [asterisk-users] University dumps CISCO VoIP for Asterisk

I suspect the article is referring to BLF, which is a traditional Key 
System feature.  It does not scale well in larger PBXs.


BLF support is not great (in Asterisk OR in phones) for SIP.

joea, j4computers wrote:
. . 


What SPECIFICALLY are you trying to do that you are unable to do?



No specifics, at this time, too early in evaluation.  

I get the point, I think, about thousands of buttons.  


My concerns are the ability to answer on multiple lines, and have various options,upon no 
pickup, including have selected (or all) unanswered calls ring thru to, or be picked up 
by, an admin or catch all operator.

I was assured this was cake for asterisk, but was concerned by the article.

joea
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Re: [asterisk-users] Tracking the source of a disconnect? - SOLVED


Jamin W. Collins wrote:

Doug Lytle wrote:

Jamin W. Collins wrote:

 callprogress = yes

The only thing I'm iffy about is the above entry.

Maybe it's mistaking the progress as disconnect?


That does appear to have been the issue.  We haven't had a new 
occurrence of the random disconnects since disabling callprogress.


The comments in /etc/asterisk/zapata.conf didn't tip you off?

;
; On trunk interfaces (FXS) it can be useful to attempt to follow the 
progress

; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
; progress attempts to determine answer, busy, and ringing on phone lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
; so don't count on it being very accurate.
;
; Few zones are supported at the time of this writing, but may be selected
; with progzone
;
; This feature can also easily detect false hangups. The symptoms of this is
; being disconnected in the middle of a call for no reason.
;
;callprogress=yes
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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread adebayo omo-dare

Hi Sheerwood,   I unfortunately saw a bit of what I percieve to be an error in what you said. BerkeleyDB does in fact support replication across nodes - see: http://www.sleepycat.com/docs/ref/rep/intro.html- possibly you meant to say the version implemented in * does not support replication. If so, I do appoligise for beinga little pedantic.I have only just started to look at *'s code - so what I say further is with a great deal of hesitation when directly referenced to *. However, I work with both Berkely (on a programming level)and MySQL in a telecom (soft-switch) environment.In terms of performance (judged as speed), a comparison between MySQL and Berkeley would be like comparing a top of the range Mercedes to an F1 racing car. Overheads from MySQL come in the form of SQL translation, use of Sockets, etc... This
 is in addition to its size.Yet, the choice between the two, is a lot more complex, IMHO, than mereley thinking in terms of performance. And possible High Availability solutions, in their own rights, taking in to consideration that * will be workingin concert with numerous otherenvironments,programmes and requirments,are diverse enough to make each deployment a little unique - thereby making each option a potential liability.One rule of thumb for us has always been - if you need raw speed, and intend to deal with the data in a very restricted/rigid/"well defined"manner - opt for Berkeley. But if you want a great deal of fluidity, and intend, or may at some time intend,for that data to interact with other applications and potential requirements -Opt MySQL.It is possibly also best to work with what you feel most comfortable with first and
 then experiment to see if you may require the services of the other.ps. In terms of querying a DB for every call, I would presume that a DB is an active and fragilething and the provision of ACID ensures that everything that occurs with it does sowith a certain measure of safety. In fact, due to the random manner of requests, you will find it, in complete terms,actually performs a lot better than any other form of retrieval.Hope this, in some manner,helps  Bayo  Rushowr [EMAIL PROTECTED] wrote:   good stuff mate.  a few clarifications: you had static "extensions.conf", realtime "sipusers", etc, right?  Also, abt features like call fwding, etc, which one is better, performance
 wise, using a mysql db, or use Asterisk's internal DB(berkeley db, isnt it?using those DBput n DBget ops)??Anyone's got any figures for these?  This somewot spoils the fun in Asterisk, when talking of performance, to query the DB for every call . Sort of pulls things down. Any comments or observations guys?  - Ben.Ben,Yes, static extensions.conf, realtime everything else. A realtimedialplan never made much sense to me, as the dialplan shouldn't (in myhumble opinion) be that fluid anyway, it should be fairly static.In terms of spoiling the fun and/or performance issues, let me note thatin my current implementation we not only have options being queried butalso realtime billing, permissions, limits, and carrier/trunkperformance data, all being pulled and calculated via the database. Ialso have handy little timers returning the length of time it takes todo the
 processing from request receipt to dial, and I'm still currentlyunder 1-2 seconds for entire call preparation including all the logicthat goes along with checking all features, the current account'saccount status, balance and limits, AND all parent accounts in it's"billing chain". I haven't done a head to head with the berkley DB, butI think part of the reason it's so fast is due to the highly normalizeddatabase structure, which allows for efficient query design. It's notall third form, but almost there :D.I'm in the last days of ALPHA now with my current project. Once welaunch BETA, which will be a semi-public testing by invitation (Murph,you still going to participate?), I should be able to find a few minutesto outline the design.One other quick thing, the berkley DB doesn't allow for clusteringeither, MySQL does. Very nice to have your database distributed acrossmultiple nodes, makes for an easier time
 designing the failovers :DCheers,Sherwood___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users 
		 
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Re: [asterisk-users] Tracking the source of a disconnect? - SOLVED

2006-09-20 Thread Jamin W. Collins


Eric ManxPower Wieling wrote:


The comments in /etc/asterisk/zapata.conf didn't tip you off?

;
; On trunk interfaces (FXS) it can be useful to attempt to follow the 
progress

; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
; progress attempts to determine answer, busy, and ringing on phone lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
; so don't count on it being very accurate.



Based on the comments, I mistakenly thought the setting would be ignored 
on non-FXS devices.  Specifically since the PRI already had the 
signaling out of band for all of this.


I thought knowing for sure that this fixed the issue I reported might be 
useful to others.  So, I reported back that it had in fact corrected it. 
 I apologize if my error has offended your sensibilities in some way.


--
Jamin W. Collins
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Re: [asterisk-users] Grandstream SX2000 attended tranfer

2006-09-20 Thread Faris Raouf


magnus wrote:

Hi all, could anyone share how to perform attended transfers with Asterisk
and Grandstream SX2000's - we are able to perform blind transfers with no
problem, but attended transfers fail - is it necessary to set two line
identities on the phones to be able to do this?
Appreciate all input, thanks - Magnus



Funny you should ask -- I was going to ask the exact same question about 
the GXP-2000 (is that the model you mean or is there a new similar 
phone?). At any rate they both seem to have the same problem:


In order to do an attended transfer on the Grandstreams we have to have 
two accounts defined on the phone (both on separate usernames/numbers in 
our case - maybe you can do it with one?), one on Line 1 and one on Line 2.


Call comes in on Line 1. Put caller on hold. Dial person you want to 
transfer to on Line 2. Then transfer.


I've tried pressing Line 2 until the identity of Line 1 comes up - i.e. 
reuse Line 1 - but this does not work. Instantly fails.


The instruction manual gives completely different instructions but these 
simply do not work.


And what is not clear is how the transfer works when using the strange 
two account situation - is the transfer going * - phone - person you 
are transferred to once transferred? (can reinvite = no incidentally) or 
is the phone


This is all completely unlike the case with a Polycom where it just lets 
you transfer with no problems and just one line.


I'm using the latest stable firmware on the Grandstreams - it has been 
like this for all firmware versions I've used for over a year now.


Faris.

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

On Wed, Sep 20, 2006 at 03:57:07PM +0100, adebayo omo-dare wrote:
 Hi Sheerwood, 
   I unfortunately saw a bit of what I percieve to be an error in what 
 you said. BerkeleyDB does in fact support replication across nodes - 
 see: http://www.sleepycat.com/docs/ref/rep/intro.html - possibly you 
 meant to say the version implemented in * does not support replication. 
 If so, I do appoligise for being a little pedantic.


The version in Asterisk is the last one before the relicense to the
Sleepycat license. 1.86 (?), and not 4.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Zap channel digit.

2006-09-20 Thread Luca Salemmi

Title: Zap channel digit.






I have a problem in outbound call.


My extension is.


exten = _0X.,1,Dial(zap/1/${EXTEN},20,TW)

exten = _0X.,2,Dial(zap/2/${EXTEN},20,TW)

exten = _0X.,3,Dial(zap/3/${EXTEN},20,TW)

exten = _0X.,4,Dial(zap/4/${EXTEN\},20,TW)

exten = _0X,105,Playback(tt-allbusy)


The problem is: when i digit 0+number if the digit is not speedy asterisk don't use the complete number.


How i can do?


Thanks Luca


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Re: [asterisk-users] two safe_asterisk processes on the same PBX???

On Fri, Sep 15, 2006 at 09:14:25AM -0500, Moises Silva wrote:
 If you want to have a safe asterisk I would recommend using svscan
 from daemontools package, more wonderfull software of D.J. Bernstein.
 
 http://cr.yp.to/daemontools/svscan.html

Assumming you really want to live with DJB-style file system.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] Re: SIP Lines Example Citel

2006-09-20 Thread Steven

Setting Realm to asterisk worked for me.

ref. from sip.conf:
;realm=mydomain.tld  ; Realm for digest authentication
; defaults to asterisk. If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name



-- 
-- 
Steven

http://www.glimasoutheast.org



Steve Totaro [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
 Anyone know how to setup the SIP lines on a Citel box so it can register with 
 Asterisk.  I keep getting Unauthorized and I have 
 tried every different combination of settings that I can think of.  I am not 
 sure what fields are required or what information 
 goes where in the Citel interface.

 Thanks,
 Steve Totaro
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Re: [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?

2006-09-20 Thread Craig Guy

I'm interested, too in how to accomplish this.  I have tried earlier today 
with a Snom360 to register it using its mac address as the authentication 
username.  I can't seem to get it to work (hopefully I'm just doing 
something wrong).


My sip.conf (asterisk 1.2.12) looks something like:

[9580]
type=peer
auth=000413242fff:[EMAIL PROTECTED]

With this the handset registers itself with asterisk, however I don't think 
it is working as I can change the username and password without affecting 
the registration on the handset.  If I try and set secret=secret, or 
md5secret= then asterisk refuses to register the handset with a 
'Registration from ... failed for ... - Username/auth name mismatch'  How 
can I specify the authentication username in sip.conf?


Craig

- Original Message - 
From: Tomislav Parčina [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Monday, September 18, 2006 2:31 PM
Subject: [asterisk-users] Re: Can you explain why multiple registration isan 
important (missing) feature ?



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

And there is your problem.  Using the extension as the SIP User ID does
not scale, is confusing, and limits your thinking about devices and
extensions.  There are several reasons this is a bad idea.  Multiple
extension numbers ringing on the same device / line appearance is the
most common.

We use the MAC address of the device as the SIP User ID.  We append a
-a, -b, -c, etc to the MAC address for each line appearance.  This does
not work well for Softphone, but since All Softphones Suck(TM), we don't
really care about this limitation.

Users seldom need to know their SIP User ID.


Can you please tell me more about this. I don't follow you weary well. I 
understand that we need to treat phone and users different, but I don't 
thing that is easy to do with Asterisk 1.2. Maybe something will change, but 
till then...




--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Tracking the source of a disconnect? - SOLVED