date:20060209

On Thu, Feb 09, 2006 at 09:12:44AM +, Pete Barnwell wrote:
 On Wed, 2006-02-08 at 12:58 -0500, Matt Roth wrote:
  Keep in mind that if you want to run Asterisk Business Edition, RedHat 
  Enterprise 3 or Fedora Core 3 are currently required in order to receive 
  full technical support.  My options were narrowed down further by the 
  amount of RAM in our production server.  It has 20GBs, and all of the 
  documentation for RHEL3 mentioned limits below that.  I don't know if 
  those are hard limits or tech support limits, but either way it made the 
  choice to use FC3 obvious.
 
 ES is limited to 16Gb. AS doesn't have a limit mentioned anywhere,
 except to use the 'hugemem' kernel  16Gb

I'll tell you a little secret (nobody is listening, right?)

ES and AS are of the same codebase. IIRC even the same kernel and same
everything. The only difference is the license.

So I expect CentOS not to be limited this way.

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

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Re: [Asterisk-Users] Polycom dialplan restriction

2006-02-09 Thread Doug Lytle



Carlos Chavez wrote:

 Is there any way to increase the number of digits before the number is
diales automatically?
  


Yes,

I don't know about the 601s, but under the 301s and the 501s you can 
edit the digit map via the web interface or the sip.cfg on your ftp server.


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] 4 TE411P in one server installation

2006-02-09 Thread Raymond Chen









Dear all,



Does anyone try to install 2 or multiple TE411 card into
one server? Can it be done? What about stability?



Thanks



Ray






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RE: [Asterisk-Users] Leading 0 on caller ID with internal S0 (HFC)

2006-02-09 Thread Pedro Nunes


Hi Sven,

Same problem... Not solved...
With CAPI and mISDN. 

I think it as to do with 

nationalprefix=0
internationalprefix=00

on capi.conf/misdn.conf. I already try to nationalprefix= but always
get that damn 0. If I change nationalprefix=5 I get a leading 5 and so
on... But without any leading digit I couldn't do it yet.

Pedro Nunes


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sven
Fischer
Sent: quinta-feira, 9 de Fevereiro de 2006 9:23
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Leading 0 on caller ID with internal S0 (HFC)

Hi all,

I have a problem: On my internal S0 where phones are connected via HFC I
get 
all the number with a leading 0 (either from internal SIP phones or
external 
dialins via CAPI). I don't know where to look for this 0. Any ideas?

Greetings, Sven

-- 
Sven Fischer (Dipl.-Phys.) - FACIT Consulting GmbH
  Hausinger Str. 6 - 40764 Langenfeld
  Tel: 02173/16700-55 Fax: 02173/16700-60

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Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delaybetweenanswering and ringing

2006-02-09 Thread Chris Stenton


What have you set the

PSTN Dialing Delay:

on the PSTN Line tab (logged in as admin advanced) ?

Mine is set to 1 and it works well.

Chris

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

Sent: Wednesday, February 08, 2006 9:50 PM
Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long 
delaybetweenanswering and ringing




Hi Jean-Michel,

We did actually try the 'r' option, but it has no effect, as Asterisk will 
only supply ringing until the dialed SIP extension answers, which it does 
immediately. The 4 second delay occurs between when the SPA-3000 answers 
the SIP call and then places the PSTN one. I believe that the ringing tone 
is provided by the PSTN at that point.


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Feb 8, 2006, at 1:41 PM, Jean-Michel Hiver wrote:


Anthony Rodgers a écrit :

 Greetings,

 We are currently testing a Sipura SPA-3000 as a gateway from our
 Asterisk system to a PSTN line for 911 access. We have a number of
 locations and want to place an SPA-3000 in each, connected to a PSTN
 line that will provide the correct ANI/ALI information to 911 for
each
 location.

 It all works great, except for a reasonably significant (4 seconds)
 delay between when the SPA-3000 answers the SIP call from the
Asterisk
 server (immediately upon dialing, according to the Asterisk CLI) and
 the ringing tone begins (the remote phone begins ringing at that same
 time).

 The delay is enough for users to think that the phone isn't working -
 not what you want for 911!

 Any ideas?

You could use the 'r' flag in your Dial() command to simulate a ringing
tone instantly. This is less than ideal though. Have you done some SIP
traces (using ngrep for examples) to look when the SIP 'ringing' signal
is actually being sent?

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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RE: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - longdelaybetweenanswering and ringing

2006-02-09 Thread Sam Lee

You can even set it to zero. Mine works well when in zero. The line pick up 
immediately : 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton
Sent: Thursday, February 09, 2006 6:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - 
longdelaybetweenanswering and ringing

What have you set the

PSTN Dialing Delay:

on the PSTN Line tab (logged in as admin advanced) ?

Mine is set to 1 and it works well.

Chris

- Original Message -
From: Anthony Rodgers [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Wednesday, February 08, 2006 9:50 PM
Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long 
delaybetweenanswering and ringing

 Hi Jean-Michel,

 We did actually try the 'r' option, but it has no effect, as Asterisk 
 will only supply ringing until the dialed SIP extension answers, which 
 it does immediately. The 4 second delay occurs between when the 
 SPA-3000 answers the SIP call and then places the PSTN one. I believe 
 that the ringing tone is provided by the PSTN at that point.

 Regards,
 --
 Anthony Rodgers
 Business Systems Analyst
 District of North Vancouver
 Web: http://www.dnv.org
 RSS Feed: http://www.dnv.org/rss.asp

 On Feb 8, 2006, at 1:41 PM, Jean-Michel Hiver wrote:

 Anthony Rodgers a écrit :

  Greetings,

  We are currently testing a Sipura SPA-3000 as a gateway from our 
  Asterisk system to a PSTN line for 911 access. We have a number of 
  locations and want to place an SPA-3000 in each, connected to a 
  PSTN line that will provide the correct ANI/ALI information to 911 
  for
 each
  location.

  It all works great, except for a reasonably significant (4 seconds) 
  delay between when the SPA-3000 answers the SIP call from the
 Asterisk
  server (immediately upon dialing, according to the Asterisk CLI) 
  and the ringing tone begins (the remote phone begins ringing at 
  that same time).

  The delay is enough for users to think that the phone isn't working 
  - not what you want for 911!

  Any ideas?

 You could use the 'r' flag in your Dial() command to simulate a 
 ringing tone instantly. This is less than ideal though. Have you done 
 some SIP traces (using ngrep for examples) to look when the SIP 
 'ringing' signal is actually being sent?

 Cheers,
 Jean-Michel.

 --
 Jean-Michel Hiver - http://ykoz.net/
 Découvrez la Réunion des Technologies IP  Telecom
 TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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RE: [Asterisk-Users] No sound on 10% of incoming calls

2006-02-09 Thread Jerome SOUCANY

Hello,

I changed these parameters in zapata.conf :
callprogress=no
busydetect=no

And now it's working fine.

Jerome

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jerome
SOUCANY
Envoyé : mardi 7 février 2006 11:04
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] No sound on 10% of incoming calls

Hello,
 
I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring
but I don't hear the caller and the caller doesn't hear me (all IP Phones
have the same problem).

This problem appear also if the call is directly send to the second E1 of
the digium card who is connected to an IVR.

It does not depand on the charge of the server (I have the problem with only
one call).

The configuration :
 
PRI (France Telecom) 15 channels   Asterisk = IP Phone
 
* Server : 
- Dell power edge 1800SC
- 2 Ethernet cards (LAN + VoIP LAN)
- Digium card : TE 405P
- Linux Mandriva LE 2005 (10.2) :
  Linux ASTERISK 2.6.11-12mdksmp #1 SMP i686 Intel(R) Xeon(TM) CPU
3.00GHz unknown GNU/Linux
  - Asterisk 1.2.4
- Zaptel 1.2.3
- Libpri 1.2.2

* IP Phone :
SNOM 320 (latest firmware)


zaptel.conf

span=1,1,0,ccs,hdb3
span=2,1,0,ccs,hdb3,crc4,yellow
span=3,1,0,ccs,hdb3,crc4,yellow
span=4,1,0,ccs,hdb3,crc4,yellow

bchan = 1-15, 17-31
dchan = 16
bchan = 32-46,48-62
dchan = 47
bchan = 63-77,79-93
dchan = 78
bchan = 94-108,110-124
dchan = 109

loadzone = fr
defaultzone = fr




zapata.conf

[channels]
switchtype=euroisdn
pridialplan=national
signalling=pri_cpe
usecallerid=yes
hidecallerid=yes
usecallingpres=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=-6.0

group=1
callgroup=1
pickupgroup=1

immediate=no
callprogress=yes

callerid=asreceived
group=1
context=from-pstn
signalling=pri_cpe
channel = 1-15;,17-31  = only 15 first channels on PRI

group=2
context=from-ivr
signalling=pri_net
channel = 32-46,48-62

group=3
context=from-ivr-bis
signalling=pri_net
channel = 63-77,79-93

group=4
signalling=pri_net
channel = 94-108,110-124





Any ideas ? 



Regards

Jerome


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[Asterisk-Users] Asterisk 1.2.x + ooh323 from addons - incoming call goes always to default context.

2006-02-09 Thread Jarek Jarzebowski


Hi all,

I am trying to setup h.323 connection between two asterisks. The 
situation is like that:


asterisk173 only must accept incomming h.323 calls from asterisk172, so 
asterisk173 is peer and asterisk172 is user, am I right?


My config files:

Asterisk173:


ooh323.conf:


(...)
;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
faststart=no
;h245tunneling=no

;H323-ID to be used for asterisk server
;Default - Asterisk PBX
h323id=ObjSysAsterisk
e164=100

;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk

(...)

[asterisk172]
type=user
context=asterisk172
disallow=all
allow=ulaw

(...)


Asterisk173:
=

ooh323.conf:


(...)

;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
;faststart=no
;h245tunneling=no


;H323-ID to be used for asterisk server
;Default - Asterisk PBX
;h323id=ObjSysAsterisk
h323id=asterisk172
e164=100

;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk

;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;gatekeeper = a.b.c.d
gatekeeper = DISABLE

(...)

[asterisk173]
type=peer
ip=83.142.201.173
port=1720
disallow=all
allow=ulaw

(...)

Situation is that call is coming to asterisk173 but it goes to [default] 
context not to [asterisk172]


I have no idea left why it is not going to [asterisk172].
Please give me some advice.

P.S.
What is the meaning of e164 and FastStart parameters?

Regards,
--
Jarek

--
Jarek
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Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway -longdelaybetweenanswering and ringing

2006-02-09 Thread Chris Stenton

I had problems with it set to 0 for some reason but that was a very early 
firmware for the device.

Chris

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

Sent: Thursday, February 09, 2006 11:14 AM
Subject: RE: [Asterisk-Users] SPA-3000 VOIP-PSTN 
gateway -longdelaybetweenanswering and ringing

You can even set it to zero. Mine works well when in zero. The line pick 
up immediately :

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Chris 
Stenton

Sent: Thursday, February 09, 2006 6:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - 
longdelaybetweenanswering and ringing

What have you set the

PSTN Dialing Delay:

on the PSTN Line tab (logged in as admin advanced) ?

Mine is set to 1 and it works well.

Chris

- Original Message -
From: Anthony Rodgers [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 08, 2006 9:50 PM
Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long 
delaybetweenanswering and ringing

Hi Jean-Michel,

We did actually try the 'r' option, but it has no effect, as Asterisk
will only supply ringing until the dialed SIP extension answers, which
it does immediately. The 4 second delay occurs between when the
SPA-3000 answers the SIP call and then places the PSTN one. I believe
that the ringing tone is provided by the PSTN at that point.

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp

On Feb 8, 2006, at 1:41 PM, Jean-Michel Hiver wrote:

Anthony Rodgers a écrit :

 Greetings,

 We are currently testing a Sipura SPA-3000 as a gateway from our
 Asterisk system to a PSTN line for 911 access. We have a number of
 locations and want to place an SPA-3000 in each, connected to a
 PSTN line that will provide the correct ANI/ALI information to 911
 for
each
 location.

 It all works great, except for a reasonably significant (4 seconds)
 delay between when the SPA-3000 answers the SIP call from the
Asterisk
 server (immediately upon dialing, according to the Asterisk CLI)
 and the ringing tone begins (the remote phone begins ringing at
 that same time).

 The delay is enough for users to think that the phone isn't working
 - not what you want for 911!

 Any ideas?

You could use the 'r' flag in your Dial() command to simulate a
ringing tone instantly. This is less than ideal though. Have you done
some SIP traces (using ngrep for examples) to look when the SIP
'ringing' signal is actually being sent?

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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[Asterisk-Users] Busy problem

2006-02-09 Thread Jerome SOUCANY

Hello,

I have a busy problem with Asterisk when I try to transfer a call from PRI
directly to IVR.
This problem appear sometime after 2 hours or 2 minutes.

The log file contain :
Unable to create channel of type 'Zap' (cause 34 - Circuit/channel
congestion)

When this problem appear I must restart Asterisk to solve it.

Another thing, I don't know why the alarm is set to NOP on SPAN 2 ?Maybe
it comes from here ?

Any ideas ? 

The configuration :
 
PRI (France Telecom) 15 channels   (SPAN1) Asterisk (SPAN2) =
IVR
 
Server : 
- Dell power edge 1800SC
- 2 Ethernet cards (LAN + VoIP LAN)
- Digium card : TE 405P
- Linux Mandriva LE 2005 (10.2) :
  Linux ASTERISK 2.6.11-12mdksmp #1 SMP i686 Intel(R) Xeon(TM) CPU
3.00GHz unknown GNU/Linux
  - Asterisk 1.2.4
- Zaptel 1.2.3
- Libpri 1.2.2



ASTERISK*CLI zap show status
Description  Alarms IRQbpviol
CRC4
T4XXP (PCI) Card 0 Span 1OK 0  0  0
T4XXP (PCI) Card 0 Span 2NOP0  0  0
T4XXP (PCI) Card 0 Span 3NOP0  0  0
T4XXP (PCI) Card 0 Span 4RED/NOP0  0  0

zaptel.conf

span=1,1,0,ccs,hdb3
span=2,1,0,ccs,hdb3,crc4,yellow
span=3,1,0,ccs,hdb3,crc4,yellow
span=4,1,0,ccs,hdb3,crc4,yellow

bchan = 1-15, 17-31
dchan = 16
bchan = 32-46,48-62
dchan = 47
bchan = 63-77,79-93
dchan = 78
bchan = 94-108,110-124
dchan = 109

loadzone = fr
defaultzone = fr




zapata.conf

[channels]
switchtype=euroisdn
pridialplan=national
signalling=pri_cpe
usecallerid=yes
hidecallerid=yes
usecallingpres=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=-6.0

group=1
callgroup=1
pickupgroup=1

immediate=no
callprogress=no
busydetect=no

callerid=asreceived
group=1
context=from-pstn
signalling=pri_cpe
channel = 1-15;,17-31  = only 15 first channels on PRI

group=2
context=from-ivr
signalling=pri_net
channel = 32-46,48-62

group=3
context=from-ivr-bis
signalling=pri_net
channel = 63-77,79-93

group=4
signalling=pri_net
channel = 94-108,110-124




Regards

Jerome



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[Asterisk-Users] clid and src fields wrong in cdr

2006-02-09 Thread asterisk

Hi all,
I have a strange problem, regarding zap channels and cdr.

I am using asterisk bristuffed version
Asterisk 1.2.2-BRIstuffed-0.3.0-PRE-1i, Copyright (C) 1999 - 2006 Digium,
Inc. and others.

with two billion ISDN cards. I also installed asterisk addons, last stable
version via cvs

internal calls, or calls starting from internal sip or iax phone are
recorded in the cdr all without any problem

incoming external calls, reaching as via 1 of the two billion cards, have
problems in cdr recording.
If the call reach channel 4 or 5, which are the 2 channels of the second
zap group, everything is OK
If the call reach channel 1 or 2, which are the 2 channels of the FIRST zap
group, everything is OK but src and clid, wich contain strange symbols
(example 'xƒ' ora '' and so on)

More interesting, when the call arrives on one of these 2 channels (1 or 2)
and is routed to one internal SIP Phone, the phone display
correctly shows the CallerID. So it means tha CallerID reach the * box

If I look at the full log, I can see:

Feb  9 11:00:35 DEBUG[25990] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Feb  9 11:00:35 DEBUG[25990] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
 VALUES ('2006-02-09 11:00:30','xƒ','xƒ','0108680580','custom-did-route',
'Zap/2-1','SIP/580-2357','Dial','SIP/580|25|tr',5,2,'ANSWERED',3,'')
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'xƒ'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'xƒ'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '0108680580'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'custom-did-route'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'Zap/2-1'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'SIP/580-2357'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'Dial'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'SIP/580|25|tr'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '2006-02-09
11:00:30'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '2006-02-09
11:00:33'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '2006-02-09
11:00:35'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '5'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '2'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'ANSWERED'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'DOCUMENTATION'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '(null)'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is
'asterisk-25591-1139479230.10'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '(null)'

while in the other 2 channels:
Feb  9 10:57:13 DEBUG[25689] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Feb  9 10:57:13 DEBUG[25689] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
 VALUES ('2006-02-09
10:57:08','0108680550','0108680550','0108680580','custom-did-route',
'Zap/4-1','SIP/580-1b7c','Dial','SIP/580|25|tr',5,3,'ANSWERED',3,'')
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '0108680550'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '0108680550'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '0108680580'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is 'custom-did-route'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is 'Zap/4-1'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is 'SIP/580-1b7c'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is 'Dial'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is 'SIP/580|25|tr'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '2006-02-09
10:57:08'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '2006-02-09
10:57:10'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '2006-02-09
10:57:13'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '5'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '3'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is 'ANSWERED'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is 'DOCUMENTATION'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '(null)'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is
'asterisk-25591-1139479028.6'
Feb  9 10:57:13 DEBUG[25689] pbx.c: Function result is '(null)'

I tried several configuration, all the same problem.
here are my files:
asterisk01:/etc/asterisk # cat /etc/zaptel.conf
# hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=it
defaultzone=it

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

span=2,1,3,ccs,ami
bchan=4-5
dchan=6



asterisk01:/etc/asterisk # cat /etc/asterisk/zapata.conf
;# Flash Operator Panel will parse this file for zap trunk buttons
;# AMPLABEL will be used for the display labels on the buttons

;# %c Zap Channel number
;#

Re: [Asterisk-Users] MFC/R2 in Brazil

2006-02-09 Thread Melcon Moraes


Can you send some *CLI output? BTW, which spandsp version are you using?

[]'s
MM

Darlon wrote:

I don´t know if the last message was with content. So, I sent again. I have
installed a Digium card TE210P and unicall for use MFC/R2. I think that it´s
all right but I can´t make and receive calls. I´m using asterisk 2.1 with
the patch made by José P. Leitão and the follow libs:

 libsupertone-0.0.2
 libunicall-0.0.3
 libmfcr2-0.0.3
 zaptel 2.1

My number is 34318300. The Telco send me only 8300. I see that I receive
from the Telco the first digit (8) and my asterisk answer 5, but the Telco
doesn´t receive my digit. My configs are below:
I tried to change the timer in mfcr2.c to 2. I tried a lot of
combinations in protocolvariant but, no sucess.
The Telco said me that the PBX is synchronized. Strange, no?

Please help me. Thanks a lot.


 *unicall.conf*

;call telephony channel driver
; Sample configuration file

[channels]
loglevel=255
language=br
context=default
usecallerid=yes
hidecallerid=no
restrictcid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
relaxdtmf=yes
rxgain=0.0
txgain= 0.0
callgroup=1
pickupgroup=1
immediate=no
callerid=asreceived
amaflags=default
accountcode=line-E1
faxdetect=no
musiconhold=default
protocolclass=mfcr2
protocolvariant=br,10,4
protocolend=cpe
group=1
channel = 1-15
channel = 17-31


 *zaptel.conf*

 span=1,1,0,cas,hdb3
 cas=1-15:1101
 cas=17-31:1101
 loadzone = br
 defaultzone=br


;extensions.conf*

[general]
static=yes
writeprotect=no

[default]
exten = _,1,SetCallerID(Betha Sistemas,4834318300)
exten = _,2,Dial(Unicall/g1/${EXTEN},60,t)
exten = _3XXX,1,Macro(sipiax,IAX2/${EXTEN})?

exten=8300,1,Goto(telefonista,s,1) ;ligação cai na fila da telefonista
exten=8301,1,Macro(sipiax,IAX2/3001) ;ligação cai diretamente no ramal
desejado
exten=8302,1,Goto(suporte_tributos,s,1) ;ligação cai na fila do suporte
tributos
exten=8303,1,Goto(telefonista,s,1) ;ligação cai na fila da telefonista
exten=8304,1,Goto(telefonista,s,1) ;ligação cai na fila da telefonista

;Fila de Atendimento Telefonista
[telefonista]
exten=s,1,Answer(2)
exten=s,2,SetMusicOnHold(default)
exten=s,3,Queue(telefonista)

;Fila de Atendimento Suporte Tributos
[suporte_tributos]
exten=s,1,Answer()
exten=s,2,SetMusicOnHold(default)
exten=s,3,DigitTimeout,5
exten=s,4,ResponseTimeout,10
exten=s,5,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP})
;exten=s,5,Background(fila_de_atendimento)
exten=s,6,Queue(suporte-tributos)

;Login para a fila de atendimento
exten=801,1,Wait,1
exten=801,2,AgentLogin()

[macro-sipiax]
exten=s,1,SetLanguage(${LANG})
exten=s,2,SetCallerId(${CALLERID})
exten=s,3,Dial(${ARG1},20,Ttr)
exten=s,4,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten=s-NOANSWER,2,Hangup()
exten=s-CHANUNAVAIL,1,Voicemail(u${MACRO_EXTEN}) ;O ramal está indisponível
exten=s-CHANUNAVAIL,2,Hangup()
exten=s-BUSY,1,Voicemail(b${MACRO_EXTEN});o ramal não está ocupadodo
exten=s-BUSY,2,Hangup()
exten=s-CONGESTION,1,Voicemail(b${MACRO_EXTEN});o ramal não está
´disponível
exten=s-CONGESTION,2,Hangup()


Darlon Ferreira Bortolini
Rede/Desenvolvimento
Betha Sistemas
Fone (48) 431-0750/Ramal 1000

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Re: [Asterisk-Users] IAX registration expiration

2006-02-09 Thread Vincent Régnard


Vincent Régnard wrote:

Joseph Rothstein a écrit :


I can't seem to change the default registration for IAX clients:

Feb  6 12:22:52 NOTICE[7883]: chan_iax2.c:5673 update_registry: 
Restricting

registration for peer 'virbiage' to 60 seconds (requested 3600)
Feb  6 12:23:03 NOTICE[7883]: chan_iax2.c:5673 update_registry: 
Restricting

registration for peer 'test1' to 60 seconds (requested 1200)

Can this be controlled on a peer-by-peer basis?

Thanks,
Joe



Hi,

I do have the same problem with my asterisk 1.2.4
Even if I set expire settings as follow:

;; iax.conf sample
[general]
bindport=4569
bindaddr=0.0.0.0
bandwidth=high
tos=lowdelay
minregexpire=60
maxregexpire=16000
defaultexpire=600

I get the following notices:

Feb  6 15:21:05 NOTICE[23075]: chan_iax2.c:5676 update_registry: 
Restricting registration for peer 'test' to 60 seconds (requested 600)
Feb  6 15:23:19 NOTICE[23075]: chan_iax2.c:5676 update_registry: 
Restricting registration for peer 'test2' to 60 seconds (requested 300)


I am wondering if this is a bug or if I misconfigured somewhere ?


This was due to the fact that I had two [global] sections in the config 
file. Now everything is going fine.


smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [Asterisk-Users] clid and src fields wrong in cdr SOLVED

2006-02-09 Thread asterisk

Today I noticed that junghanns released the bristuffed version of asterisk
1.2.4 (it was .1.2.2 last week, when I installed)
http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1l.tar.gz

Downloading and installing that solved my problem

Andrea


   
 [EMAIL PROTECTED] 
 .it   
 Sent by:   To 
 asterisk-users-bo asterisk-users@lists.digium.com   
 [EMAIL PROTECTED]  cc 
 m.com 
   Subject 
   [Asterisk-Users] clid and src   
 09/02/2006 12.32  fields wrong in cdr 
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




Hi all,
I have a strange problem, regarding zap channels and cdr.

I am using asterisk bristuffed version
Asterisk 1.2.2-BRIstuffed-0.3.0-PRE-1i, Copyright (C) 1999 - 2006 Digium,
Inc. and others.

with two billion ISDN cards. I also installed asterisk addons, last stable
version via cvs

internal calls, or calls starting from internal sip or iax phone are
recorded in the cdr all without any problem

incoming external calls, reaching as via 1 of the two billion cards, have
problems in cdr recording.
If the call reach channel 4 or 5, which are the 2 channels of the second
zap group, everything is OK
If the call reach channel 1 or 2, which are the 2 channels of the FIRST zap
group, everything is OK but src and clid, wich contain strange symbols
(example 'xƒ' ora '' and so on)

More interesting, when the call arrives on one of these 2 channels (1 or 2)
and is routed to one internal SIP Phone, the phone display
correctly shows the CallerID. So it means tha CallerID reach the * box

If I look at the full log, I can see:

Feb  9 11:00:35 DEBUG[25990] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Feb  9 11:00:35 DEBUG[25990] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)

 VALUES ('2006-02-09 11:00:30','xƒ','xƒ','0108680580','custom-did-route',
'Zap/2-1','SIP/580-2357','Dial','SIP/580|25|tr',5,2,'ANSWERED',3,'')
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'xƒ'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'xƒ'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '0108680580'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'custom-did-route'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'Zap/2-1'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'SIP/580-2357'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'Dial'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'SIP/580|25|tr'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '2006-02-09
11:00:30'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '2006-02-09
11:00:33'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '2006-02-09
11:00:35'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '5'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '2'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'ANSWERED'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is 'DOCUMENTATION'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '(null)'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is
'asterisk-25591-1139479230.10'
Feb  9 11:00:35 DEBUG[25990] pbx.c: Function result is '(null)'

while in the other 2 channels:
Feb  9 10:57:13 DEBUG[25689] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Feb  9 10:57:13 DEBUG[25689] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)

 VALUES ('2006-02-09
10:57:08','0108680550','0108680550','0108680580','custom-did-route',

[Asterisk-Users] Queue transfer

2006-02-09 Thread Tomislav Parčina

When I try to make att transfer (*2) of call that was in queue the call get's 
disconnected. Blind transfer (#1) works fine. In dial plan I don't have any h 
or H (hangup call with *). In features.conf I have this line disconnect = *0.

What could be the reason why call hang's up?


--
Tomislav Parcina
tparcina#lama.hr

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Re: [Asterisk-Users] Queue - check agent

2006-02-09 Thread Joel Vandal


Hi,


I have defined 4 queue's. Is there any way to check is there any agent logged 
in any of those queue's?

What I would like to do is to check if there is any agent in any of queue's and 
if there is, then I'll will transfer a call to that queue, it there isn't I 
would like to do something else with a call.
 



The Queue application sets the QUEUESTATUS channel variable upon 
completion.  The status of the call can be :  TIMEOUT, FULL, JOINEMPTY, 
LEAVEEMPTY, JOINUNAVAIL or LEAVEUNAVAIL.


Here an example

...
exten   = 3,5,Queue(scopserv-test|tH|||30)
exten   = 3,6,GotoIf($[${QUEUESTATUS} = JOINEMPTY]?1000)
exten   = 3,7,GotoIf($[${QUEUESTATUS} = JOINUNAVAIL]?1000)
exten   = 3,8,GotoIf($[${QUEUESTATUS} = FULL]?1000)
exten   = 3,9,NoOp(Normal Queue exist)
exten   = 3,10,Hangup

exten= 3,1000,Voicemail([EMAIL PROTECTED])
   
--

Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com/
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[Asterisk-Users] asterisk logger - urgent!!!




Hi,

Since yesterday my Asterisk 1.2.3 is displaying the 
following message every few seconds

Asterisk Event Logger restartedRotated 
Logs Per SIGXFSZ (Exceeded file size limit)

This causes my log files (verbose, queue_log) to 
become huge with lots of logger rotate messages, but I don't know which files is 
exceeding size limit, since even if I delete all log files I still get this 
message.

Any way, I have plenty of disk space and couldn't 
find the reason for this message.

Please help me identify the 
issue!Dov


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Re: [Asterisk-Users] Asterisk native sounds now available!

2006-02-09 Thread Mark Phillips


Yes, it seems that I was somewhat in error.

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


kevin ling wrote:

In my remember, when playback a file. The Asterisk will automatically choose
the audio file with the lowest conversion cost. Not always looks the
filename.gsm. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Thursday, February 09, 2006 5:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk native sounds now available!

Yes you can copy them into the same directory as the current files. Kris
recommends that you move your existing files for safety only.

The mode (ULAW, GSM etc) is selected by Asterisk depending upon what mode
the current caller is using.

Have you noticed that you don't have to put a file extension on the end of a
Playback instruction? This is because Asterisk looks for filename.mode when
trying to play a file. In the event it can't find filename.mode it looks for
filename.gsm.

If the file it's playing is not encoded using the current mode it has to
transcode the gsm file into whatever is required. This not only adds
computing overhead to the call in progress but degrades the quality of the
file as all such transactions are lossy.

Understand?

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com




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[Asterisk-Users] Dell PowerEdge 1800 and TE410P

2006-02-09 Thread Niall Hallett

Hi,

I've been asked to find a server that is capable of running the Digium
TE410P card. We usually get Dell PE servers and after a quick look I
think the PE 1800 has the required slot:

Six Total: 2 PCI Express (x8 lane  x4 lane); 2 x 64-bit/100MHz PCI-X; 1
x 32-bit/33MHz PCI (5v) and 1 x 64-bit/66MHz PCI

Has anyone got this hardware, does it work with the card and asterisk?

Thanks,
Niall


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RE: [Asterisk-Users] Dell PowerEdge 1800 and TE410P

2006-02-09 Thread Jerome SOUCANY

Hello,

I use this hardware but I have some problems and I don't if these problems
come from the DELL or not.

http://www.digium.com/index.php?menu=compatibility

Regards

Jerome 

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Niall Hallett
Envoyé : jeudi 9 février 2006 14:11
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Dell PowerEdge 1800 and TE410P

Hi,

I've been asked to find a server that is capable of running the Digium
TE410P card. We usually get Dell PE servers and after a quick look I think
the PE 1800 has the required slot:

Six Total: 2 PCI Express (x8 lane  x4 lane); 2 x 64-bit/100MHz PCI-X; 1 x
32-bit/33MHz PCI (5v) and 1 x 64-bit/66MHz PCI

Has anyone got this hardware, does it work with the card and asterisk?

Thanks,
Niall


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[Asterisk-Users] Re: asterisk logger - urgent!!!




I found the problem.

Master.csv reached 2.0GB and since the moment this 
happened Asterisk went crazy!

Since I am using cdr-mysql, how do I disable the 
use of csvs?

Thank you
Dov

  - Original Message - 
  From: 
  Dov Bigio 

  To: asterisk-users@lists.digium.com 
  
  Sent: Thursday, February 09, 2006 10:56 
  AM
  Subject: asterisk logger - 
urgent!!!
  
  Hi,
  
  Since yesterday my Asterisk 1.2.3 is displaying 
  the following message every few seconds
  
  Asterisk Event Logger 
  restartedRotated Logs Per SIGXFSZ (Exceeded file size 
  limit)
  
  This causes my log files (verbose, queue_log) to 
  become huge with lots of logger rotate messages, but I don't know which files 
  is exceeding size limit, since even if I delete all log files I still get this 
  message.
  
  Any way, I have plenty of disk space and couldn't 
  find the reason for this message.
  
  Please help me identify the 
  issue!Dov
  
  
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[Asterisk-Users] Question on SIP authentication with users from OpenSER

2006-02-09 Thread Barry Flanagan


Hi,

We are using Asterisk 1.2.3 with RealTime for PSTN and Voicemail where 
users register with an OpenSER cluster (2 nodes currently).


When they request PSTN they are forwarded to * where they have entries 
in SIP realtime database. This ensures that they get their correct 
CallerID and context, etc.


This is working fine at present, where I have the SIP users set up with 
the following relevant SIP entries:


  name  username
  callerid  User XXX
   canreinvite  no
   context  context
  dtmfmode  RFC2833
  host  87.232.1.16
  insecure  port
  type  friend
  username  username


Note that I have set the host to the IP of the OpenSER server, and there 
is no secret.


I have the OpenSER servers set up as peers also.

My questions are:

1. Is this the best way to to set this up?

2. I have many users, and I need to be certain that a) the username 
exists and b) that the request came from one of our OpenSER servers. 
Will the above ensure that both the username AND the host are correct? I 
have seen instances where if I have a static SIP entry with the same 
host= line, a non-existent user will be accepted as this static user.


3. How can have more than one possible host= setting for a user (i.e. 
they could come in from either of our OpenSER servers.



Thanks!

--

-Barry Flanagan
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Re: [Asterisk-Users] Bandwidth: to seperate or not to seperate

2006-02-09 Thread Rich Adamson

inline...

 You said:
 Help us understand exactly what this incoming traffic flooding the
 bandwidth is suppose to mean. Are you running something else besides web
 and voip through this link?  If not, then what is flooding your 
 bandwidth?
 
 You are right about web page serving not using much incoming bandwidth (good 
 for one-sided QoS 
management).  I was inaccurate by saying hosting webpages - we also have all 
email traffic and 
host racked servers for customers of ours too (and I have them on a lower QoS 
in the 
Netfilter/Wondershaper setup).  Actually, typically, the servers are business 
customers and 
probably don't use much bandwidth at all but I can't be sure that one of them 
would not, at any 
time, upload data from their mega-high-speed office connection - its a bit of 
an unknown.
 

The above comments would be of some concern, particularly if some sends an email
to the server with a large attachment (or whatever). Given all the other 
traffic,
you're faced with a choice to micro-manage the existing bandwidth, or, do as you
mentioned providing two paths.

Some time ago, someone on the list suggested a QoS-like app (maybe it was 
wondershaper,
don't remember) that does impact inbound traffic. My understanding is the app 
delays
TCP response packets (from your server to the external user) essentially 
slowing the
inbound flow of traffic. If you think about how TCP functions, an ACK packet is
required after approximately three inbound packets acknowledging the receipt of
those three packets; if the ACK packet is delayed by xxx milliseconds, it 
essentially
impacts the speed at which incoming packets arrive. No such thing for UDP 
traffic
though. Since web and email traffic uses TCP, that might be something to look 
into.

 It's a bit of a bummer that inbound traffic shaping cannot be done - 
 considering that its a 
data-center setup as opposed to an office/home setup (kind of a 
so-close-but-so-far type thing).
 

True inbound packet shaping would actually require the sender to prioritize 
all
packets, and assumes every layer-2 and layer-3 device between the sender and 
your
hardware respect QoS settings. That's not going to happen anytime soon, although
some Internet providers do in fact respect it.

 Thinking about it now, before paying money for something we don't need, I 
 should probably try to 
graph bandwidth usage (which opens a can of worms too - I'm used to MRTG but it 
would only show a 
5 minute average and not instant peaks that would affect VoIP quality - have 
you ever used any 
other graphing tools?).
 

You might try STG to graph the usage. Its available everywhere on the Internet 
and
can be set to poll at one second intervals if that's actually necessary. I use
it a lot with five or ten second polling to see peaks, etc.

Rich


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Re: [Asterisk-Users] asterisk logger - urgent!!!

On Thu, Feb 09, 2006 at 10:56:18AM -0200, Dov Bigio wrote:
 Hi,
 
 Since yesterday my Asterisk 1.2.3 is displaying the following message every 
 few seconds
 
 Asterisk Event Logger restarted
 Rotated Logs Per SIGXFSZ (Exceeded file size limit)
 
 This causes my log files (verbose, queue_log) to become huge with lots 
 of logger rotate messages, but I don't know which files is exceeding 
 size limit, since even if I delete all log files I still get this message.

Unrelated to the origin of the problem:

Do you run 'logger reload' after deleting those logs? Otherwise Asterisk
still writes to the old (deleted) logs

 
 Any way, I have plenty of disk space and couldn't find the reason for this 
 message.
 
 Please help me identify the issue!
 Dov
 

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

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Re: [Asterisk-Users] Free IAX login

2006-02-09 Thread Asterisk guy

[guest]
type=friend
context=default
isecure=very


-it doesn;t work ,  asterisk shows:  Feb  9 08:41:13
NOTICE[29683]: chan_iax2.c:6782 socket_read: Rejected connect attempt
from 89.*.8...
for the incoming call






On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote:
 Try adding insecure=very to the guest user account in iax.conf. This
 should not do a user/pass challenge on the incoming call.

 Mark, G7LTT/KC2ENI
 Randolph, NJ
 http://www.g7ltt.com


 kevin ling wrote:
  Not sure answer your question? Try to write some html code and let user
  register the username  password online.
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
  Sent: Tuesday, February 07, 2006 7:31 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Free IAX login
 
  how to set up  iax.conf  , so IAX clients with any user name and any secret
  can login to * ?  ( no authorize for login )
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Re: [Asterisk-Users] Free IAX login

2006-02-09 Thread Asterisk guy

[guest]
type=friend
context=default
insecure=very


-it doesn;t work ,  asterisk shows:  Feb  9 08:41:13
NOTICE[29683]: chan_iax2.c:6782 socket_read: Rejected connect attempt
from 89.*.8...
for the incoming call

On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote:
 Try adding insecure=very to the guest user account in iax.conf. This
 should not do a user/pass challenge on the incoming call.

 Mark, G7LTT/KC2ENI
 Randolph, NJ
 http://www.g7ltt.com


 kevin ling wrote:
  Not sure answer your question? Try to write some html code and let user
  register the username  password online.
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
  Sent: Tuesday, February 07, 2006 7:31 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Free IAX login
 
  how to set up  iax.conf  , so IAX clients with any user name and any secret
  can login to * ?  ( no authorize for login )
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Re: [Asterisk-Users] Free IAX login

2006-02-09 Thread Asterisk guy

for sip.conf ,  there is a configure option for this :  allowguest=yes

is there a silimiar setting for IAX ?




On 2/7/06, Mark Phillips [EMAIL PROTECTED] wrote:
 Try adding insecure=very to the guest user account in iax.conf. This
 should not do a user/pass challenge on the incoming call.

 Mark, G7LTT/KC2ENI
 Randolph, NJ
 http://www.g7ltt.com


 kevin ling wrote:
  Not sure answer your question? Try to write some html code and let user
  register the username  password online.
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
  Sent: Tuesday, February 07, 2006 7:31 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Free IAX login
 
  how to set up  iax.conf  , so IAX clients with any user name and any secret
  can login to * ?  ( no authorize for login )
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Re: [Asterisk-Users] asterisk logger - urgent!!!

Hi Tzafrir,

The problem was the file Master.csv that had reached 2.0GB.
I am writing a cron script to backup this file periodically and prevent this
from happening.

Any way, if any developers are reading this, I don't think that rotating
asterisk logs is the best way to handle this problem!
Maybe a more user-friendly message could be logged, infoming which file
reached the 2.0GB.

About your question, yes I do, for log files. Is logger rotate could also
after I delete csv files?

Thank you
Dov

- Original Message - 
From: Tzafrir Cohen [EMAIL PROTECTED]
To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Thursday, February 09, 2006 11:42 AM
Subject: Re: [Asterisk-Users] asterisk logger - urgent!!!


 On Thu, Feb 09, 2006 at 10:56:18AM -0200, Dov Bigio wrote:
  Hi,
 
  Since yesterday my Asterisk 1.2.3 is displaying the following message
every few seconds
 
  Asterisk Event Logger restarted
  Rotated Logs Per SIGXFSZ (Exceeded file size limit)
 
  This causes my log files (verbose, queue_log) to become huge with lots
  of logger rotate messages, but I don't know which files is exceeding
  size limit, since even if I delete all log files I still get this
message.

 Unrelated to the origin of the problem:

 Do you run 'logger reload' after deleting those logs? Otherwise Asterisk
 still writes to the old (deleted) logs

 
  Any way, I have plenty of disk space and couldn't find the reason for
this message.
 
  Please help me identify the issue!
  Dov
 

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





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RE: [Asterisk-Users] Re: asterisk logger - urgent!!!

2006-02-09 Thread Allan Gee




Why 
don't you log rotate them? or if you do you should do it more 
often.

Regards Allan GeePhone: +27 21 4644400 Ext. 
103www.equation.co.za

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Dov 
  BigioSent: 09 February 2006 03:26 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: 
  asterisk logger - urgent!!!
  I found the problem.
  
  Master.csv reached 2.0GB and since the moment 
  this happened Asterisk went crazy!
  
  Since I am using cdr-mysql, how do I disable the 
  use of csvs?
  
  Thank you
  Dov
  
- Original Message - 
From: 
Dov Bigio 

To: asterisk-users@lists.digium.com 

Sent: Thursday, February 09, 2006 10:56 
AM
Subject: asterisk logger - 
urgent!!!

Hi,

Since yesterday my Asterisk 1.2.3 is displaying 
the following message every few seconds

Asterisk Event Logger 
restartedRotated Logs Per SIGXFSZ (Exceeded file size 
limit)

This causes my log files (verbose, queue_log) 
to become huge with lots of logger rotate messages, but I don't know which 
files is exceeding size limit, since even if I delete all log files I still 
get this message.

Any way, I have plenty of disk space and 
couldn't find the reason for this message.

Please help me identify the 
issue!Dov


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RE: [Asterisk-Users] Queue - check agent

2006-02-09 Thread David Waugh

Hello,
I might be wrong here, but I thought that in Queues.conf, if you defined a 
queue with joinempty=no, or joinempty=strict then no calls will be placed in 
the queue, and asterisk will go onto the next extension in the dial plan.

; This setting controls whether callers can join a queue with no members. There
; are three choices:
;
; yes- callers can join a queue with no members or only unavailable members
; no - callers cannot join a queue with no members
; strict - callers cannot join a queue with no members or only unavailable
;  members
;
; joinempty = yes
;
; If you wish to remove callers from the queue when new callers cannot join,
; set this setting to one of the same choices for 'joinempty'
;
; leavewhenempty = yes


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joel Vandal
Sent: 09 February 2006 12:55
To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Queue - check agent


Hi,

I have defined 4 queue's. Is there any way to check is there any agent logged 
in any of those queue's?

What I would like to do is to check if there is any agent in any of queue's 
and if there is, then I'll will transfer a call to that queue, it there isn't 
I would like to do something else with a call.
  


The Queue application sets the QUEUESTATUS channel variable upon 
completion.  The status of the call can be :  TIMEOUT, FULL, JOINEMPTY, 
LEAVEEMPTY, JOINUNAVAIL or LEAVEUNAVAIL.

Here an example

...
exten   = 3,5,Queue(scopserv-test|tH|||30)
exten   = 3,6,GotoIf($[${QUEUESTATUS} = JOINEMPTY]?1000)
exten   = 3,7,GotoIf($[${QUEUESTATUS} = JOINUNAVAIL]?1000)
exten   = 3,8,GotoIf($[${QUEUESTATUS} = FULL]?1000)
exten   = 3,9,NoOp(Normal Queue exist)
exten   = 3,10,Hangup

exten= 3,1000,Voicemail([EMAIL PROTECTED])

--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com/
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RE: [Asterisk-Users] Cisco 2620 as PRI gateway

2006-02-09 Thread Tim Reimers




Yeah-- sorry...
"
dial-peer voice 635099 voipdescription calls sent 
to Asteriskpreference 1destination-pattern 
[635-9]..progress_ind setup enable 3session target 
ipv4:10.10.1.28dtmf-relay h245-alphanumeric
"

I had been trying to do this with H.323 -- the Call Manager 
uses H.323

There are some sip commands available in that dial-peer 

ACS-GW(config-dial-peer)#voice-class sip ? 
rel1xx Type of reliable provisional response 
support transport Configure transport related 
parameters url url type in 
request line of outgoing INVITE

Not sure how I set those---

This:
voice-class codec 1voice-class h323 
1
is what is in there for the Call Manager h.323 dial-peer 


That's obviously NOT what I want for the Asterisk-SIP 
connection... 

but I don't know what Ineed to do regarding the 'sip 
url' or 'sip transport' or 'sip rel1xx' commands, if 
anything...

How does one debug SIP activity? I see the debugs for 
calls--- but I don't know the related debugs for actively 
watching--
like you would 'debug isdn q931' -- that's the 
outgoing side of the router--
what would be the debug for a SIP call 'arriving' at the 
router??





From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Juan 
SalasSent: Wednesday, February 08, 2006 2:17 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] Cisco 2620 as PRI gateway

Did 
you create the dial-peers in the2651?


  -Mensaje original-De: Tim Reimers 
  [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, 
  February 08, 2006 1:41 PMPara: Asterisk Users Mailing List - 
  Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] Cisco 2620 as 
  PRI gateway
  sip-ua sip-server ipv4:asterisk server 
  ip address
  OK -
  So I added those lines to my 2651 with the IP of my 
  asterisk box...
  
  How would I set this up as a SIP trunk in 
  Asterisk?
  I have done this, in building a SIP trunk in 
  AMP.
  
  host=10.12.1.252type=friend
  
  I 
  don't know if/how to specify a username/password (as was the defaults in 
  there- the router didn't support having that configured..)
  So I 
  picked friend..
  
  Then, in call routing, I picked my "Outbound 
  Routing"
  the 
  "9_outside" route of "9|."
  Set 
  that to use the new 'gw-rtr' I'd created...
  
  no 
  go...
  
  Debug ISDN q931 doesn't show anything going to the 
  router...
  
  In 
  Asterisk- 
  " -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" 
  back from 10.12.1.252"
  snipped from below
  
  The 
  router doesn't show anything...
  
  
  
  
  the below shows up in 
  Asterisk - mode
  -- Executing Macro("SIP/6351-cc18", 
  "dialout-trunk|3|2439499|") in new stack -- Executing 
  GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto 
  (macro-dialout-trunk,s,3) -- Executing 
  Macro("SIP/6351-cc18", "user-callerid") in new stack -- 
  Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new 
  stack -- DBget: varname=AMPUSER, family=DEVICE, 
  key=6351/user -- DBget: set variable AMPUSER to 
  6351 -- Executing DBget("SIP/6351-cc18", 
  "AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack -- 
  DBget: varname=AMPUSERCIDNAME, family=AMPUSER, 
  key=6351/cidname -- DBget: set variable AMPUSERCIDNAME 
  to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", "0?5") 
  in new stack -- Executing SetCallerID("SIP/6351-cc18", 
  "Tim-Zyxel 6351") in new stack -- Executing 
  NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" 6351") in new 
  stack -- Executing Macro("SIP/6351-cc18", 
  "record-enable|6351|OUT") in new stack -- Executing 
  GotoIf("SIP/6351-cc18", "0  0?2:4") in new stack -- 
  Goto (macro-record-enable,s,4) -- Executing 
  AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new 
  stack -- Launched AGI Script 
  /var/lib/asterisk/agi-bin/recordingcheck 
  recordingcheck|20060208-115748|1139417868.14: Outbound recording not 
  enabled -- AGI Script recordingcheck completed, 
  returning 0 -- Executing NoOp("SIP/6351-cc18", "No 
  recording needed") in new stack -- Executing 
  Macro("SIP/6351-cc18", "outbound-callerid|3") in new 
  stack -- Executing GotoIf("SIP/6351-cc18", "1?3") in new 
  stack -- Goto 
  (macro-outbound-callerid,s,3) -- Executing 
  DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new 
  stack -- DBget: varname=USEROUTCID, family=AMPUSER, 
  key=6351/outboundcid -- DBget: set variable USEROUTCID 
  to 6351 -- Executing GotoIf("SIP/6351-cc18", "0?6") in 
  new stack -- Executing SetCallerID("SIP/6351-cc18", 
  "6351") in new stack -- Executing NoOp("SIP/6351-cc18", 
  "CallerID set to 6351") in new stack -- Executing 
  SetGroup("SIP/6351-cc18", "OUT_3") in new stack -- 
  Executing CheckGroup("SIP/6351-cc18", "") in new stack 
  -- Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499") in new 
  stack -- Executing SetVar("SIP/6351-cc18", 
  "DIAL_TRUNK=3") in new stack -- Executing 
  AGI("SIP/6351-cc18", "fixlocalprefix") in new stack -- 
  Launched AGI Script

[Asterisk-Users] Re: Remapping Polycom IP501 buttons

2006-02-09 Thread Noah Miller

Hi Henry -

 Just started using an asterisk-based PBX with Polycom IP501 phones.  Am
 Fairly satisfied and am starting to get into FTP setup of the phones.
 Have figured out most things except for how button remapping works.
 
 In sip.cfg, I have this entry:
 
keys key.IP_500.31.function.prim=DoNotDisturb/keys
 
 This works as expected but if I try to change the remapping to any other
 value like MyStatus, SpeedDialMenu, or BuddyStatus, it doesn't work.
  I got the list of values from Polycom's admin guide.  Why does
 DoNotDisturb work and no other values that I've tried?

You've run into the same problem a lot of other people have had.  Remapping
hard keys works fine, but remapping soft keys does not.  In fact, trying to
remap the soft keys results in some pretty weird behavior.  The Polycom
manual is a little misleading in that it doesn't mention this at all.  My
best guess is that the softkeys don't work because they can mean different
things depending on what the phone is doing at the time.  Polycom, if you're
reading this, this would be another great feature to have!

- Noah

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[Asterisk-Users] Asterisk with Billing

2006-02-09 Thread ram

Hi

i would like to start with prepaid and post paid billing system

i would like to have your feed back what all i need to look
and what is the best billing software

where i can configure DID incoming also

some suggestions areapprciated

ram
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Re: [Asterisk-Users] ztdummy on gentoo 2005.1

2006-02-09 Thread Sean Cook

Check your /etc/modules.d/zaptel and make sure you have:

install ztdummy /sbin/modprobe --ignore-install ztdummy  /sbin/ztcfg

Sean

Tzafrir Cohen wrote:

On Wed, Feb 08, 2006 at 07:10:16PM -0600, Miguel wrote:
  

Hi i followed this instructions for installing ztdummy on a 2.6 kernel 
(taken from 
http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy

)

cd /usr/src/zaptel

   * READ /usr/src/zaptel/README.udev and follow the steps
   * check modules on: /etc/sysconfig/zaptel if you have no digium
 hardware comment out all modeules except ztdummy.

- make linux26
- make install
- Reboot to make udev changes take effect



Why reboot? Shouldn't it take effect immidietly for new devices?

  

- modprobe ztdummy

i dont have any digium card so i deleted all but ztdummy from 
/etc/modules.d/zaptel (in gentoo is not /etc/sysconfig/zaptel).
all seems ok after the reboot, but when i run

modprobe ztdummy

this is what i get:

voip zaptel-1.2.3 # modprobe ztdummy
FATAL: Error inserting ztdummy 
(/lib/modules/2.6.12-gentoo-r10/misc/ztdummy.o): Invalid module format
FATAL: Error running install command for ztdummy



ztdummy.o shouldn't have been copied in the first place, though.

Could you please give a more detailed log of what happened in 'make
linux26' and 'make install'?

e.g: 

make clean
make linux26 install 21 | tee log

  


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[Asterisk-Users] Asterisk vs. Traditional PBX

2006-02-09 Thread Nora Lavelle

Title: Asterisk vs. Traditional PBX

Hi everyone !

So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have an asterisk deployment with about 15 people on it. We have sipura 841 phones. The biggest issue currently is voice quality. lot of complaints there. I have a dell 650 poweredge (single processory system), with a digium tdm400 card and 4 analog lines plugged into it.

So here are my questions:

* Is asterisk a good solution for my company ? or should I just install the traditional pbx and look to move to asterisk in a couple of years ? (I personally would prefer asterisk cuz I'm a unix person not a phone person so from a manageability perspective i would love this )

* If I were to go to an asterisk solution to support about 200 people with the requirements above what hardware platform would you recommend ? I'm guessing I'd need a PRI line and a different digium card? Also would a 1cpu poweredge dell be enough ? or would that have to be upgraded too ?

If anyone is running an environment similar to this that can provide help I would really appreciate this. I'm having a hard time making this decision and would love to hear anybody's experience in a real time environment.

Thanks again this list ROCKS!
Nora Lavelle

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RE: [Asterisk-Users] Dell PowerEdge 1800 and TE410P

2006-02-09 Thread Niall Hallett

Yeah, I saw that compatibility list and the potential problem with the
onboard ethernet controller. Did you disable it and use a pci based card
instead?

Does anyone else run PowerEdge servers with the TE410P?

Thanks,
Niall

On Thu, 2006-02-09 at 14:24 +0100, Jerome SOUCANY wrote:
 Hello,
 
 I use this hardware but I have some problems and I don't if these problems
 come from the DELL or not.
 
 http://www.digium.com/index.php?menu=compatibility
 
 Regards
 
 Jerome 
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de Niall Hallett
 Envoyé : jeudi 9 février 2006 14:11
 À : asterisk-users@lists.digium.com
 Objet : [Asterisk-Users] Dell PowerEdge 1800 and TE410P
 
 Hi,
 
 I've been asked to find a server that is capable of running the Digium
 TE410P card. We usually get Dell PE servers and after a quick look I think
 the PE 1800 has the required slot:
 
 Six Total: 2 PCI Express (x8 lane  x4 lane); 2 x 64-bit/100MHz PCI-X; 1 x
 32-bit/33MHz PCI (5v) and 1 x 64-bit/66MHz PCI
 
 Has anyone got this hardware, does it work with the card and asterisk?
 
 Thanks,
 Niall
 
 
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RE: [Asterisk-Users] Cisco 2620 as PRI gateway

2006-02-09 Thread Juan Salas




If you 
are using the 2620 like a SIP IP-PSTN gateway
your 
voip dial-peer would be like this:

dial-peer voice 
635099 voipdescription calls sent to Asteriskpreference 
1destination-patternT 
(or whatever youneed to match)session 
targetsip-serverdtmf-relay h245-alphanumeric (or 
whatever you need)
session-protocol sip
no vad 


And you need a pots 
dial-peer,
something like this

dial-peer voice 0 potsdestination-pattern 
T (or whatever you need)port 
0/0 0
And in sip-ua:

sip-ua sip-server asterisk server ip 
address
This is the basic


Regards

Jsalas



  -Mensaje original-De: Tim Reimers 
  [mailto:[EMAIL PROTECTED]Enviado el: Thursday, 
  February 09, 2006 10:04 AMPara: Asterisk Users Mailing List - 
  Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] Cisco 2620 as 
  PRI gateway
  Yeah-- sorry...
  "
  dial-peer voice 635099 voipdescription calls 
  sent to Asteriskpreference 1destination-pattern 
  [635-9]..progress_ind setup enable 3session target 
  ipv4:10.10.1.28dtmf-relay h245-alphanumeric
  "
  
  I had been trying to do this with H.323 -- the Call 
  Manager uses H.323
  
  There are some sip commands available in that dial-peer 
  
  ACS-GW(config-dial-peer)#voice-class sip ? 
  rel1xx Type of reliable provisional response 
  support transport Configure transport related 
  parameters url url type in 
  request line of outgoing INVITE
  
  Not sure how I set those---
  
  This:
  voice-class codec 1voice-class h323 
  1
  is what is in there for the Call Manager h.323 dial-peer 
  
  
  That's obviously NOT what I want for the Asterisk-SIP 
  connection... 
  
  but I don't know what Ineed to do regarding the 
  'sip url' or 'sip transport' or 'sip rel1xx' commands, if 
  anything...
  
  How does one debug SIP activity? I see the debugs for 
  calls--- but I don't know the related debugs for actively 
  watching--
  like you would 'debug isdn q931' -- that's the 
  outgoing side of the router--
  what would be the debug for a SIP call 'arriving' at the 
  router??
  
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Juan 
  SalasSent: Wednesday, February 08, 2006 2:17 PMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  RE: [Asterisk-Users] Cisco 2620 as PRI gateway
  
  Did 
  you create the dial-peers in the2651?
  
  
-Mensaje original-De: Tim Reimers 
[mailto:[EMAIL PROTECTED]Enviado el: Wednesday, 
February 08, 2006 1:41 PMPara: Asterisk Users Mailing List - 
Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] Cisco 2620 
as PRI gateway
sip-ua sip-server ipv4:asterisk server 
ip address
OK -
So I added those lines to my 2651 with the IP of my 
asterisk box...

How would I set this up as a SIP trunk in 
Asterisk?
I have done this, in building a SIP trunk in 
AMP.

host=10.12.1.252type=friend

I 
don't know if/how to specify a username/password (as was the defaults in 
there- the router didn't support having that 
configured..)
So 
I picked friend..

Then, in call routing, I picked my "Outbound 
Routing"
the "9_outside" route of "9|."
Set that to use the new 'gw-rtr' I'd created...

no 
go...

Debug ISDN q931 doesn't show anything going to the 
router...

In 
Asterisk- 
" -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" 
back from 10.12.1.252"
snipped from below

The router doesn't show anything...




the below 
shows up in Asterisk - mode
-- Executing Macro("SIP/6351-cc18", 
"dialout-trunk|3|2439499|") in new stack -- Executing 
GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto 
(macro-dialout-trunk,s,3) -- Executing 
Macro("SIP/6351-cc18", "user-callerid") in new stack 
-- Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new 
stack -- DBget: varname=AMPUSER, family=DEVICE, 
key=6351/user -- DBget: set variable AMPUSER to 
6351 -- Executing DBget("SIP/6351-cc18", 
"AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack -- 
DBget: varname=AMPUSERCIDNAME, family=AMPUSER, 
key=6351/cidname -- DBget: set variable AMPUSERCIDNAME 
to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", 
"0?5") in new stack -- Executing 
SetCallerID("SIP/6351-cc18", "Tim-Zyxel 6351") in new 
stack -- Executing NoOp("SIP/6351-cc18", "Using 
CallerID "Tim-Zyxel" 6351") in new stack -- 
Executing Macro("SIP/6351-cc18", "record-enable|6351|OUT") in new 
stack -- Executing GotoIf("SIP/6351-cc18", "0  
0?2:4") in new stack -- Goto 
(macro-record-enable,s,4) -- Executing 
AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new 
stack -- Launched AGI Script 
/var/lib/asterisk/agi-bin/recordingcheck 
recordingcheck|20060208-115748|1139417868.14: Outbound recording not 
enabled -- AGI Script

RE: [Asterisk-Users] more cpu intensive echo cancellers ?

2006-02-09 Thread Gerard Saraber

I appreciate the input, but after doing a little research on that card,
it looks like I'll still need the channel bank, I think with some
carefull ebaying, I should be able to do the hardware canceling for
about $1000 less then what i saw the 104d card for.
not to mention it seems total overkill, I've got a wimpy 10 pstn phone
lines, a quad T1 card seems a little excessive to me. If there was a
single T1 with the G.168 echo canceler card for say $800, I'd be all
over that (still researching).

I've got all this extra cpu power, and nothing to use it on ;)
in the mean time, I'll put the 104d card on the list of possibilities,

Thanks,
Gerard Saraber
[EMAIL PROTECTED]



On Wed, 2006-02-08 at 17:26 -0800, Canuck15 wrote:
 Gerard,
 
 Just get yourself a Sangoma card with hardware echo can and be done with it.
 It is worth every penny just for the headaches it will save you.  It's a
 better solution for most situations compared to a channel bank.  Cheaper,
 simpler and works just as good IMHO.
 
 -Original Message-
 From: Gerard Saraber [mailto:[EMAIL PROTECTED] 
 Sent: Wednesday, February 08, 2006 12:20 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] more cpu intensive echo cancellers ?
 
 Thanks for the quick reply :)
 It happens when we call a lot of different people, and obviously doesn't
 happen with our old analog phone system, so even if its caused by someone
 else, *we* still have to fix it.
 we're kind of weighing our options, I'm hoping to take care of this with
 some fancy software, but if not we'll be going the hardware canceller route.
 
 Thanks,
 Gerard Saraber
 [EMAIL PROTECTED]
 
 On Wed, 2006-02-08 at 11:45 -0800, Michael Collins wrote:
  Gerard,
  
  I'll bet your side is working great for echo cancellation.  It sounds 
  like the equipment at the other end of the call might need some help.
  You know the old rule if you and I are talking on the phone: If I hear 
  echo, you've got a problem; if you hear echo, I've got a problem.  If 
  only all echo problems were so easy to diagnose!  In any case, is it 
  possible that some of the echo you're hearing is being caused by poor 
  echo handling on the other end of the line?
  
  Just a thought.
  
  -MC
  
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Gerard 
  Saraber
  Sent: Wednesday, February 08, 2006 11:03 AM
  To: Asterisk Users Mailing List
  Subject: [Asterisk-Users] more cpu intensive echo cancellers ?
  
  Hi,
  I've had some decent luck with the mark3 echo canceller from the 
  zaptel driver, echos on about 20% of the calls, people I've called say 
  I sound great now, but our side hears echos.
  I was wondering if there was any way to tweak the current software 
  cancelers into using more CPU (and hopefully doing a better job, close 
  to a hardware canceler), I only have 10 lines, and a single call takes 
  0.5% cpu, I would have no problem if it went up to 5-10% if they would 
  work better.
  Or should I just give up now and buy the channel bank, tellabs 
  hardware echo canceler and a T1 pci card? (hope TDM400P cards have 
  decent resale value ;)
  
 --
 Regards,
 Gerard Saraber
 Network Admin, Rarcoa, Inc.
 (630) 654-2580 x11
 [EMAIL PROTECTED]
 
 
 
-- 
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]

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RE: [Asterisk-Users] Asterisk vs. Traditional PBX

2006-02-09 Thread Chad Osmond

Title: Asterisk vs. Traditional PBX



You may be using less then ideal phones. With a Polycom 501, I can't see 
you having voice quality issues, With a Sangoma or Digium card and a PRI the 
quality and functions of a Asterisk system are on par with most PBX's (I'd say 
they're above).

It is a good solution for most companies, consider the ability to change 
features and expand only limited by your abilities (or those of 
consultants).

For 200 people, you will probably need40 channels, which will be 
two T1's, so start looking for a dual T1 card ( again Digium and Sangoma make 
excellent products).

Hope this helps, there are thousands of systems running in companies of 
your size.

I 
would recommend running two servers in a active/passive format and rsync them 
every hour (to a different directory). If the server blows up and kills the 
board you can easily switch over in a few seconds.

It also makes upgrading easier,.

Chad


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Nora 
LavelleSent: February 9, 2006 9:15 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk vs. 
Traditional PBX

Hi everyone !So here's my question of the day ! I 
need to make a decision on whether or not to go to a voip solution or configure 
an existing pbx (norstar) that my company has available. We are a small 
startup. I'm wanting a solution that will support up to about 200 people, with 
direct dial-in capability, up to about 30 concurrent phone calls and good voice 
quality. Right now I have an asterisk deployment with about 15 people on it. We 
have sipura 841 phones. The biggest issue currently is voice quality. lot of 
complaints there. I have a dell 650 poweredge (single processory system), 
with a digium tdm400 card and 4 analog lines plugged into it.So here are 
my questions:* Is asterisk a good solution for my company ? or should I 
just install the traditional pbx and look to move to asterisk in a couple of 
years ? (I personally would prefer asterisk cuz I'm a unix person not a 
phone person so from a manageability perspective i would love this )* If 
I were to go to an asterisk solution to support about 200 people with the 
requirements above what hardware platform would you recommend ? I'm 
guessing I'd need a PRI line and a different digium card? Also would a 1cpu 
poweredge dell be enough ? or would that have to be upgraded too 
?If anyone is running an environment similar to this that can 
provide help I would really appreciate this. I'm having a hard time making this 
decision and would love to hear anybody's experience in a real time 
environment.Thanks again this list ROCKS!Nora 
Lavelle
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[Asterisk-Users] h323 configuration

2006-02-09 Thread Patricio Ku


Hi, Can anyone help me out setting oh323.

extensions.conf
[default]

;Prueba oh323 saliente por timanfaya0h323
exten=_6.,1,Dial(OH323/timanfayaoh323/x${EXTEN},90,tr)

[discriminador]
exten=932289394,1,SetCallerID(932289394)
exten=932289394,2,Dial(OH323/timanfayaoh323/x${EXTEN2},60,tr)

[h323-timanfaya]
exten=_6.,1,SetCallerID(12345)
exten=_6.,2,Dial(OH323/[EMAIL PROTECTED],60,tr)

oh323.conf
[general]
listenAddress=x.x.x.x (my ip)
listenPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=yes
h245inSetup=no
inBandDTMF=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
userInputMode=TONE
amaFlags=default
accountCode=H323
context=timanfayaoh323
type=user
host=x.x.x.x (provider's ip)

[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
alias=asterisk
alias=123

;alias=h323-timanfaya
alias=timanfayaoh323
alias=932289394
alias=12345
alias=10746
alias=31561

codec=G711A
frames=20
codec=G711U
frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
codec=G729
frames=8




I am getting the following error when I make a call:

-- Accepting voice call from '932289394' to '666448240' on channel 0/19, 
span 1

-- Executing SetVar(Zap/19-1, EXTEN2=666448240) in new stack
-- Executing Goto(Zap/19-1, discriminador|932289394|1) in new stack
-- Goto (discriminador,932289394,1)
-- Executing SetCallerID(Zap/19-1, 932289394) in new stack
-- Executing Dial(Zap/19-1, OH323/timanfayaoh323/19105666448240|60|tr) 
in new stack

-- H.323 call to timanfayaoh323/x666448240 with codec(s) alaw g729
Outbound H.323 call 'ip$localhost/31573'.
-- Called timanfayaoh323/x666448240
Call 'ip$localhost/31573' cleared.
-- H.323 call 'ip$localhost/31573' cleared, reason 10 (Gatekeeper cleared 
call)

Call 'ip$localhost/31573' cleared in INIT state.
-- OH323/L31573 is circuit-busy
-- Hungup 'OH323/L31573'
== Everyone is busy/congested at this time
Call 'ip$localhost/31573' without owner has already been cleared (2).
-- Channel 0/19, span 1 got hangup
-- Hungup 'Zap/19-1'

It rings 3 times and then silence.

Note, the ip and prefix are hiden for security reasons.

anyone has a clue

Sincerelly,
Patricio Ku

_
Un amor, una aventura, compañía para un viaje. Regístrate gratis en MSN Amor 
 Amistad. http://match.msn.es/match/mt.cfm?pg=channeltcid=162349


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RE: [Asterisk-Users] Nortel Meridian Opt 81C and PRI

2006-02-09 Thread Greg Camp

 -Original Message-
 From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, February 08, 2006 12:17 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Nortel Meridian Opt 81C and PRI

 On Feb 8, 2006, at 9:27 AM, Greg Camp wrote:

  Now, our latest two issues:

  1) When a user on the Nortel makes a call to a user on * a 10-digit
  callerid value shows up on the SIP phone instead of the users
  extension.
  Has anyone encountered this and found a work-around?  It's been
  suggested that we use a QSIG interface instead of 5ESS emulation, but
  did not purchased the Nortel QSIG option so it is unavailable.

 We implemented a macro that stripped the leading 6 digits from the
 numbers, like this:

 ; If it looks like one of ours, only show the last 4 digits
 exten = s,40,GotoIf($[${CALLERIDNUM:0:8} = 60498131]?50:)
 exten = s,50,SetCallerID,${CALLERIDNAME} ${CALLERIDNUM:-4}

I thought about that, but all of our extensions display the same caller id 
value.  Unfortunately the above won't work in our case.

  2) We would like to use Comedian Mail for company wide voicemail.  I
  can
  setup user extensions easily enough.  I have also setup two 4-digit
  extensions; one for picking up voicemail and one for leaving voicemail
  for an arbitrary user.  The second ext is used primarily by the
  receptionist (coming from the Nortel PBX) to redirect callers to users
  voicemails.  The issue I'm having is that if you don't pass an
  extension
  to the Voicemail() function * will prompt you one time.  If you key the
  ext incorrectly the system hangs up on you.  Is there a way to prompt
  the caller for the extension to leave a message for, accept the ext,
  check the database, and give the caller another chance if the ext is
  invalid?

 AFAIK, Voicemail() will jump to n+101 if the requested mailbox doesn't
 exist - you can use that to return to the prompt asking for the mailbox
 number.

Doh!  I completely forgot about jumping to n+101.  I have the system playing 
back invalid extension and it works like I'd hoped it would.

Thanks!
Greg

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[Asterisk-Users] Asterisk Native Sounds re-release

2006-02-09 Thread Kristian Kielhofner


Hello everyone,

	It seems that the letter s did not make it into the original release. 
 Please visit www.astlinux.org and download the latest tarball.Or, 
if you just want s in all of the available formats, just grab this:


http://mirror.astlinux.org/sounds/s.tar.bz2

Sorry!

--
Kristian Kielhofner
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RE: [Asterisk-Users] Asterisk vs. Traditional PBX

2006-02-09 Thread Kerry Garrison

Title: Asterisk vs. Traditional PBX



Hi everyone !So here's my question 
of the day ! I need to make a decision on whether or not to go to a voip 
solution or configure an existing pbx (norstar) that my company has 
available. We are a small startup. I'm wanting a solution that will 
support up to about 200 people, with direct dial-in capability, up to about 30 
concurrent phone calls and good voice quality. Right now I have an asterisk 
deployment with about 15 people on it. We have sipura 841 phones. The biggest 
issue currently is voice quality. lot of complaints there. I have a dell 
650 poweredge (single processory system), with a digium tdm400 card and 4 analog 
lines plugged into it.
[Kerry Garrison 
sayeth]
The 841 isfine for testing but I would never put 
one on a clients desk. The sound quality is bottom of the barrel. Combine that 
with the TDM400 card and itsa wonder anyone will use the phone system at 
all. Move up to the Linksys SPA941 or SPA942 or the Polycom 501and then 
use a different interface such as the Mediatrix 1204 ora PRI and your 
users will be singing your praises till the end of 
time.So here are my questions:* Is asterisk 
a good solution for my company ? or should I just install the traditional pbx 
and look to move to asterisk in a couple of years ? (I personally would prefer 
asterisk cuz I'm a unix person not a phone person so from a manageability 
perspective i would love this )[Kerry Garrison 
sayeth]
Asterisk is a great solution for your company and you 
will have many more benefits than the Northstar system. 
* If I were to go to an asterisk 
solution to support about 200 people with the requirements above what hardware 
platform would you recommend ? I'm guessing I'd need a PRI line and a 
different digium card? Also would a 1cpu poweredge dell be enough ? or would 
that have to be upgraded too ?
[Kerry Garrison 
sayeth]
You would want a beefier machine and at least one PRI. 
Its not the number of people, its the number of concurrent phone calls. I see 
businesses with 100 people and they average 5-7 concurrent calls and I have 
clients with 15 people that average 12-15 concurrent calls. 
If anyone is running an environment similar to this 
that can provide help I would really appreciate this. I'm having a hard time 
making this decision and would love to hear anybody's experience in a real time 
environment.
[Kerry Garrison 
sayeth]
My largest install is approaching 55 users, with the 
PRI and Polycom 501's they couldnt be happier. The system is on a nice 2.8ghz 
XEON system with 2gb of RAM and at peak times the server is basically 
idle.Thanks again this list ROCKS!Nora 
Lavelle
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Re: [Asterisk-Users] What ATA should I buy?

2006-02-09 Thread Michael Sampson

I've used the spa-1001 and the spa-2001 for faxes. Works good over a 
local area network. thevoipconnection sells those for about 60 bucks though.


Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000



Tomislav Parčina wrote:


I have running * without any Digium (or any other) hardware. Now I need to 
connect analog FAX machine to it. I think that cheapest and easiest way is to 
buy ATA. Please correct me if I'm wrong.

Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. 


One more question, can I plug two lines in any of those ATA-s?

Sipura SPA-2100 SIP-ATA 160$
Sipura SPA-1001 SIP-ATA 125$
ALL7902 IP SIP ATA Adapter / Router 106$
Grandstream HandyTone ATA486142$


Thank you for any suggestions.


P.S.
If this is second time you see this message, then sorry for resending, but I 
didn't see it on list...


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [Asterisk-Users] Re: asterisk logger - urgent!!!

2006-02-09 Thread Olivier Perrin

Hi, 
You just have to remove cdr_csv.so module :

unload cdr_csv.so under the CLI

or add noload = cdr_csv.so in /etc/asterisk/modules.conf and reload
asterisk



--
http://www.olivier-perrin.net


Le jeudi 09 février 2006 à 11:26 -0200, Dov Bigio a écrit :
 I found the problem.
  
 Master.csv reached 2.0GB and since the moment this happened Asterisk
 went crazy!
  
 Since I am using cdr-mysql, how do I disable the use of csvs?
  
 Thank you
 Dov
 - Original Message - 
 From: Dov Bigio 
 To: asterisk-users@lists.digium.com 
 Sent: Thursday, February 09, 2006 10:56 AM
 Subject: asterisk logger - urgent!!!
 
 
 Hi,
  
 Since yesterday my Asterisk 1.2.3 is displaying the following
 message every few seconds
  
 Asterisk Event Logger restarted
 Rotated Logs Per SIGXFSZ (Exceeded file size limit)
  
 This causes my log files (verbose, queue_log) to become huge
 with lots of logger rotate messages, but I don't know which
 files is exceeding size limit, since even if I delete all log
 files I still get this message.
  
 Any way, I have plenty of disk space and couldn't find the
 reason for this message.
  
 Please help me identify the issue!
 Dov
  
  
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Re: [Asterisk-Users] ztdummy on gentoo 2005.1

2006-02-09 Thread Miguel


Tzafrir Cohen wrote:


On Wed, Feb 08, 2006 at 07:10:16PM -0600, Miguel wrote:
 

Hi i followed this instructions for installing ztdummy on a 2.6 kernel 
(taken from 
http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy


)

cd /usr/src/zaptel

  * READ /usr/src/zaptel/README.udev and follow the steps
  * check modules on: /etc/sysconfig/zaptel if you have no digium
hardware comment out all modeules except ztdummy.

- make linux26
- make install
- Reboot to make udev changes take effect
   



Why reboot? Shouldn't it take effect immidietly for new devices?
 


I dont know, i just followed the instructions :-)

 


voip zaptel-1.2.3 # modprobe ztdummy
FATAL: Error inserting ztdummy 
(/lib/modules/2.6.12-gentoo-r10/misc/ztdummy.o): Invalid module format

FATAL: Error running install command for ztdummy
   



ztdummy.o shouldn't have been copied in the first place, though.

Could you please give a more detailed log of what happened in 'make
linux26' and 'make install'?

 



sure, this the log result:

Makefile:204: target `ztdummy.o' given more than once in the same rule.
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o gendigits.o gendigits.c

cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c   -o makefw

./makefw tormenta2.rbt tor2fw  tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw  radfw.h
Loaded 42096 bytes from file
ZAPTELVERSION= build_tools/make_version_h  version.h.tmp
if cmp -s version.h.tmp version.h ; then echo; else \
   mv version.h.tmp version.h ; \
fi

rm -f version.h.tmp
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo 
zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo 
tonezone.c

ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o torisatool.o torisatool.c

cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztmonitor.o ztmonitor.c

cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c   -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
-DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o fxotune.o fxotune.c

cc -o fxotune fxotune.o -lm
/lib/modules/2.6.12-gentoo-r10/build
make -C /lib/modules/2.6.12-gentoo-r10/build 
SUBDIRS=/usr/local/src/zaptel-1.2.3 modules

make[1]: Entering directory `/usr/src/linux-2.6.12-gentoo-r10'
/usr/local/src/zaptel-1.2.3/Makefile:204: target `ztdummy.o' given more 
than once in the same rule.

 CC [M]  /usr/local/src/zaptel-1.2.3/ztdummy.o
 Building modules, stage 2.
 MODPOST
*** Warning: zt_transmit [/usr/local/src/zaptel-1.2.3/ztdummy.ko] 
undefined!
*** Warning: zt_receive [/usr/local/src/zaptel-1.2.3/ztdummy.ko] 
undefined!
*** Warning: zt_unregister [/usr/local/src/zaptel-1.2.3/ztdummy.ko] 
undefined!
*** Warning: zt_register [/usr/local/src/zaptel-1.2.3/ztdummy.ko] 
undefined!

 CC  /usr/local/src/zaptel-1.2.3/ztdummy.mod.o
 LD [M]  /usr/local/src/zaptel-1.2.3/ztdummy.ko
make[1]: Leaving directory `/usr/src/linux-2.6.12-gentoo-r10'
cc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so 
zonedata.lo tonezone.lo

install -m 444 udev/zaptel.rules-combined /etc/udev/rules.d/zaptel.rules
install -D -m 755 ztcfg /sbin/ztcfg
if [ -f sethdlc-new ]; then \
   install -D -m 755 sethdlc-new /sbin/sethdlc; \
elif [ -f sethdlc ]; then \
   install -D -m 755 sethdlc /sbin/sethdlc ; \
fi
if [ -f zttool ]; then install -D -m 755 zttool /sbin/zttool; fi
if [ -f zaptel.ko ]; then \
   for x in ztdummy.ko ztdummy.ko; do \
   rm -f /lib/modules/2.6.12-gentoo-r10/extra/$x ; \
   done; \
   make -C /lib/modules/2.6.12-gentoo-r10/build 
SUBDIRS=/usr/local/src/zaptel-1.2.3 INSTALL_MOD_PATH= 
INSTALL_MOD_DIR=misc modules_install; \

   if ! [ -f wcfxsusb.ko ]; then \
   rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxsusb.ko; \
   fi; \
   rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxs.ko; \
else \
   for x in ztdummy.o ztdummy.o; do \
   install -D -m 644 $x /lib/modules/2.6.12-gentoo-r10/misc/$x ; \
   done; \
   if ! [ -f wcfxsusb.o ]; then \
   rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxsusb.o; \
   fi; \
   rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxs.o; \
fi
install -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0
[ `id -u` = 0 ]  /sbin/ldconfig || :
rm

[Asterisk-Users] Sip One way audio

2006-02-09 Thread Paul Oster

I've got a telecommuter working out of her home office, using a Snom 200 phone, what happens is occassionally her phone will loose audio one way.She will be talking on a call that was incoming to her extension, and all of a sudden the caller can not hear her any longer, she can here the caller fine, this has happened with both Sip-Sip calls, and calls that have come in over our PSTN circuits. The really odd thing is while troubleshooting with her yesterday I was using the one way audio to talk to her and do some packet captures, and she was using an instant message client to communicate back to me, but after being in the call for a while (didn't note exact times) the audio came back.
At first I thought this was a nat issue, and she is using Bellsouth DSL, so I had her change the dsl modem so it shares its IP address with the phone. Restarting the phone results in the phone getting the public IP address assigned via DHCP. This did not solve the issue. I've experimented with the nat settings, and the canreinvite settings but haven't had much sucess so far. I have suspicions that the cut-outs might be occuring either after the phone has been registered for a certain amount of time (possibly 1 hour) or when she has been talking for a certain amount of time (possibly 5 minutes), I'm not certain of that behavior so it may be a red herring further use of the phone will allow me to firm up if either of those statements is true.
Any suggestions would be greatly appreciated!Thank YouPaul M. OsterHere are the relevant portions of my sip.conf file...[general]port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind tocontext = incomingcall ; Default for incoming callstos=lowdelaydisallow=allallow=alawallow=gsmallow=ulaw
[104]accountcode=vsllctype=friendcontext=employeeusername=104secret=**redacted**host=dynamicqualify=yesreinvite=nocanreinvite=no[EMAIL PROTECTED],[EMAIL PROTECTED]
callgroup=1pickupgroup=1dtmfmode=rfc2833;nat=no
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[Asterisk-Users] Codec negotiation

2006-02-09 Thread Ronald Voermans




Hi 
All,

I've set up an 
Asterisk box with a SIP trunk to our PSTN provider. I've configured two incoming 
phonenumbers. One phonenumber is for voice-calls, the other one for receiving 
faxes. I want the incoming voice-calls to be coded by the G.729 codec, and the 
fax-number by G.711. Can I make a codec-negotation based on the called 
number?

If you need more 
info on this, i can send it to you.

Thank you all for 
your answer(s)!

Regards,

Ronald 
Voermans
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Re: [Asterisk-Users] Dell PowerEdge 1800 and TE410P

2006-02-09 Thread Andres


Niall Hallett wrote:


Yeah, I saw that compatibility list and the potential problem with the
onboard ethernet controller. Did you disable it and use a pci based card
instead?

Does anyone else run PowerEdge servers with the TE410P?

Thanks,
Niall

 

We run the TE410 on the PowerEdge 1850 and it is rock solid.  We do not 
have problems with the onboard ethernet controller.


--
Andres
Technical Support
http://www.telesip.net


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Re: [Asterisk-Users] Asterisk vs. Traditional PBX

2006-02-09 Thread William

Kerry is right on.

We use a similar config in dozens of installs with 200 users and it just 
cruises. Consider adding a duplicate server for failover at some point.

-Original Message-
From: Kerry Garrison [EMAIL PROTECTED]
Date: Thu, 9 Feb 2006 06:57:32 
To:'Asterisk Users Mailing List - Non-Commercial 
Discussion'asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Asterisk vs. Traditional PBX

Hi everyone !

So here's my question of the day !  I need to make a decision on whether or not 
to go to a voip solution or configure an existing pbx (norstar) that my company 
has available.  We are a small startup. I'm wanting a solution that will 
support up to about 200 people, with direct dial-in capability, up to about 30 
concurrent phone calls and good voice quality. Right now I have an asterisk 
deployment with about 15 people on it. We have sipura 841 phones. The biggest 
issue currently is voice quality. lot of complaints there.  I have a dell 650 
poweredge (single processory system), with a digium tdm400 card and 4 analog 
lines plugged into it.

[Kerry Garrison sayeth] 
The 841 is fine for testing but I would never put one on a clients desk. The 
sound quality is bottom of the barrel. Combine that with the TDM400 card and 
its a wonder anyone will use the phone system at all. Move up to the Linksys 
SPA941 or SPA942 or the Polycom 501 and then use a different interface such as 
the Mediatrix 1204 or a PRI and your users will be singing your praises till 
the end of time. 

So here are my questions:

* Is asterisk a good solution for my company ? or should I just install the 
traditional pbx and look to move to asterisk in a couple of years ? (I 
personally would prefer asterisk cuz I'm a  unix person not a phone person so 
from a manageability perspective i would love this )

[Kerry Garrison sayeth] 
Asterisk is a great solution for your company and you will have many more 
benefits than the Northstar system. 

* If I were to go to an asterisk solution to support about 200 people with the 
requirements above what hardware platform would you recommend ?  I'm guessing 
I'd need a PRI line and a different digium card? Also would a 1cpu poweredge 
dell be enough ? or would that have to be upgraded too ? 

[Kerry Garrison sayeth] 
You would want a beefier machine and at least one PRI. Its not the number of 
people, its the number of concurrent phone calls. I see businesses with 100 
people and they average 5-7 concurrent calls and I have clients with 15 people 
that average 12-15 concurrent calls.  

If anyone is running an environment similar to this that can provide help I 
would really appreciate this. I'm having a hard time making this decision and 
would love to hear anybody's experience in a real time environment.

[Kerry Garrison sayeth] 
My largest install is approaching 55 users, with the PRI and Polycom 501's they 
couldnt be happier. The system is on a nice 2.8ghz XEON system with 2gb of RAM 
and at peak times the server is basically idle. 

Thanks again this list ROCKS!
Nora Lavelle

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Sent from my BlackBerry - please excuse any typos.
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[Asterisk-Users] stable ISDN BRI card for asterisk

2006-02-09 Thread Peng Yong


we would like to running 4 port or 8 port ISDN BRI card on production asterisk
system.

any one can recommend a good product? is it stable and with good voice
quality?


-- 
Peng Yong

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RE: [Asterisk-Users] TDMoE

2006-02-09 Thread Mike Hammett

Well, I don't know what it is at the moment, I just know its a wireless T-1 
that I'd migrate over to a different infrastructure.


Actually, TDMoE can route and can go longer distances when you run it over 
Mikrotik and use their EoIP.  Well, given that the fact that it runs over 
Ethernet instead of IP is its only issue.




Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com


- Original Message - 
From: [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Thursday, February 09, 2006 3:11 AM
Subject: Asterisk-Users Digest, Vol 19, Issue 59



Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
[EMAIL PROTECTED]

You can reach the person managing the list at
[EMAIL PROTECTED]

When replying, please edit your Subject line so it is more specific
than Re: Contents of Asterisk-Users digest...


Today's Topics:

  1. Re: Asterisk-Users Digest, Vol 19, Issue 58 (Mike Hammett)
  2. RE: Re: Asterisk-Users Digest, Vol 19, Issue 58 (Alexander Lopez)
  3. RE: Two Lines, Two Businesses (Les Bell)
  4. Re: Welltech USA? and Wellgate Products? (Dinesh Nair)
  5. Re: ([EMAIL PROTECTED])
  6. Re: Two Lines, Two Businesses ([EMAIL PROTECTED])
  7. NSLU2 Asterisk (sukrit)
  8. What ATA should I buy? (Tomislav Par?ina)
  9. Queue - joinempty (Tomislav Par?ina)
 10. RE: Two Lines, Two Businesses (Alexander Lopez)
 11. Fax transmission interrupt on ISDN network (Olivier Krief)
 12. Voicemail Problem (Sam Lee)
 13. Re: ztdummy on gentoo 2005.1 (Tzafrir Cohen)
 14. Voicemailmain() refusing connection problem (Sam Lee)
 15. Tormenta 2 and channel bank (Viktor Tatianin)
 16. TDM400p (Hans Witvliet)
 17. Re: Web based SIP client (Klaus Darilion)
 18. How can I send DTMF from the console? (Anthony Azzopardi)
 19. RE: cisco 7940 firmware upgrade (kevin ling)
 20. Re: Bandwidth: to seperate or not to seperate (Derek Conniffe)
 21. RE: festival-script.pl... howto change language? (kevin ling)


--

Message: 1
Date: Thu, 9 Feb 2006 00:20:14 -0600
From: Mike Hammett [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 19, Issue 58
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; format=flowed; charset=iso-8859-1;
reply-type=original

Reason I ask is I may have a non-voice T-1 replacement project going on 
and

I'm investigating my various options.  Costs may be about the same for
turn-key and DIY.



Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com


- Original Message - 
From: [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Thursday, February 09, 2006 12:07 AM
Subject: Asterisk-Users Digest, Vol 19, Issue 58



Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
[EMAIL PROTECTED]

You can reach the person managing the list at
[EMAIL PROTECTED]

When replying, please edit your Subject line so it is more specific
than Re: Contents of Asterisk-Users digest...


Today's Topics:

  1. RE: Welltech USA? and Wellgate Products? (kevin ling)
  2. RE: Connecting to live calls (Wai Wu)
  3. RE: Web based SIP client (kevin ling)
  4. Re: 911 and ISDN PRI (Darren Nickerson)
  5. Asterisk returning 403 Forbidden response
 ([EMAIL PROTECTED])
  6. RE: Connecting to live calls (Alexander Lopez)
  7. TDMoE (Mike Hammett)
  8. SIP-H323 Help and Multiple Listening Port (Kenige Ho)
  9. RE: TDMoE (Alexander Lopez)
 10. Re: Mitel 5220 IP phones (tracinet)
 11. Polycom dialplan restriction (Carlos Chavez)
 12. SER + Asterisk (Nick Hoffman)
 13. OOH323 Configuration (Abdul Lateef)
 14. Re: Bandwidth: to seperate or not to seperate (Rich Adamson)
 15. RE: PRI indications. (Mark Edwards)


--

Message: 9
Date: Wed, 8 Feb 2006 23:59:18 -0500
From: Alexander Lopez [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] TDMoE
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

TDM is not limited to voice. But there are better ways of moving data
across an ethernet segment.

Look at the various treads recently about TDMoE.

Make sure you are using a separate card for anytype of non-testing load.
Use a 2.6 based kernel, Better networking.
Pick a religion and follow it, you with need a bit a divine
intervention.





From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Hammett
Sent: Wednesday, February 08, 2006 11:43 PM
To: asterisk-users@lists.digium.com
Subject:

Re: [Asterisk-Users] ztdummy on gentoo 2005.1

2006-02-09 Thread Miguel


Sean Cook wrote:


Check your /etc/modules.d/zaptel and make sure you have:

install ztdummy /sbin/modprobe --ignore-install ztdummy  /sbin/ztcfg

Sean

 


Mine has:


post-install ztdummy /sbin/ztcfg


I changed the line with your sugestion but same result (after reboot):

voip # lsmod
Module  Size  Used by
voip # modprobe ztdummy
FATAL: Error inserting ztdummy 
(/lib/modules/2.6.12-gentoo-r10/misc/ztdummy.o): Invalid module format

FATAL: Error running install command for ztdummy
voip mmiranda #

---
Miguel
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Re: [Asterisk-Users] ztdummy on gentoo 2005.1

On Thu, Feb 09, 2006 at 09:09:14AM -0500, Sean Cook wrote:
 Check your /etc/modules.d/zaptel and make sure you have:
 
 install ztdummy /sbin/modprobe --ignore-install ztdummy  /sbin/ztcfg

And what is that good for? You don't need to run ztcfg after loading
ztdummy.

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

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Re: [Asterisk-Users] ztdummy on gentoo 2005.1

On Thu, Feb 09, 2006 at 09:10:26AM -0600, Miguel wrote:
 Tzafrir Cohen wrote:
 
 On Wed, Feb 08, 2006 at 07:10:16PM -0600, Miguel wrote:
  
 
 Hi i followed this instructions for installing ztdummy on a 2.6 kernel 
 (taken from 
 http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy
 
 )
 
 cd /usr/src/zaptel
 
   * READ /usr/src/zaptel/README.udev and follow the steps
   * check modules on: /etc/sysconfig/zaptel if you have no digium
 hardware comment out all modeules except ztdummy.
 
 - make linux26
 - make install
 - Reboot to make udev changes take effect

 
 
 Why reboot? Shouldn't it take effect immidietly for new devices?
  
 
 I dont know, i just followed the instructions :-)
 
  
 
 voip zaptel-1.2.3 # modprobe ztdummy
 FATAL: Error inserting ztdummy 
 (/lib/modules/2.6.12-gentoo-r10/misc/ztdummy.o): Invalid module format
 FATAL: Error running install command for ztdummy

 
 
 ztdummy.o shouldn't have been copied in the first place, though.
 
 Could you please give a more detailed log of what happened in 'make
 linux26' and 'make install'?
 
  
 
 
 sure, this the log result:
 
 Makefile:204: target `ztdummy.o' given more than once in the same rule.

Something is bad. Did you edit Makefile?

 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o gendigits.o gendigits.c
 cc -o gendigits gendigits.o -lm
 ./gendigits
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c   -o makefw
 ./makefw tormenta2.rbt tor2fw  tor2fw.h
 Loaded 69900 bytes from file
 ./makefw pciradio.rbt radfw  radfw.h
 Loaded 42096 bytes from file
 ZAPTELVERSION= build_tools/make_version_h  version.h.tmp
 if cmp -s version.h.tmp version.h ; then echo; else \
mv version.h.tmp version.h ; \
 fi
 
 rm -f version.h.tmp
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztcfg.o ztcfg.c
 cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo 
 zonedata.c
 cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo 
 tonezone.c
 ar rcs libtonezone.a zonedata.lo tonezone.lo
 cc -o ztcfg ztcfg.o libtonezone.a -lm
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o torisatool.o torisatool.c
 cc -o torisatool torisatool.o
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o ztmonitor.o ztmonitor.c
 cc -o ztmonitor ztmonitor.o
 cc -o ztspeed.o -c ztspeed.c
 cc -o ztspeed ztspeed.o
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c   -o zttest
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA 
 -DZAPTEL_CONFIG=\/etc/zaptel.conf\   -c -o fxotune.o fxotune.c
 cc -o fxotune fxotune.o -lm
 /lib/modules/2.6.12-gentoo-r10/build
 make -C /lib/modules/2.6.12-gentoo-r10/build 
 SUBDIRS=/usr/local/src/zaptel-1.2.3 modules
 make[1]: Entering directory `/usr/src/linux-2.6.12-gentoo-r10'
 /usr/local/src/zaptel-1.2.3/Makefile:204: target `ztdummy.o' given more 
 than once in the same rule.
  CC [M]  /usr/local/src/zaptel-1.2.3/ztdummy.o
  Building modules, stage 2.
  MODPOST
 *** Warning: zt_transmit [/usr/local/src/zaptel-1.2.3/ztdummy.ko] 
 undefined!
 *** Warning: zt_receive [/usr/local/src/zaptel-1.2.3/ztdummy.ko] 
 undefined!
 *** Warning: zt_unregister [/usr/local/src/zaptel-1.2.3/ztdummy.ko] 
 undefined!
 *** Warning: zt_register [/usr/local/src/zaptel-1.2.3/ztdummy.ko] 
 undefined!

Something is wrong. Not sure exactly what.

Now the overly-complicated install target's script springs into action
and manages to install the .o file into the modules dir. Funny, don't
you think?

  CC  /usr/local/src/zaptel-1.2.3/ztdummy.mod.o
  LD [M]  /usr/local/src/zaptel-1.2.3/ztdummy.ko
 make[1]: Leaving directory `/usr/src/linux-2.6.12-gentoo-r10'
 cc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so 
 zonedata.lo tonezone.lo
 install -m 444 udev/zaptel.rules-combined /etc/udev/rules.d/zaptel.rules
 install -D -m 755 ztcfg /sbin/ztcfg
 if [ -f sethdlc-new ]; then \
install -D -m 755 sethdlc-new /sbin/sethdlc; \
 elif [ -f sethdlc ]; then \
install -D -m 755 sethdlc /sbin/sethdlc ; \
 fi
 if [ -f zttool ]; then install -D -m 755 zttool /sbin/zttool; fi
 if [ -f zaptel.ko ]; then \
for x in ztdummy.ko ztdummy.ko; do \
rm -f /lib/modules/2.6.12-gentoo-r10/extra/$x ; \
done; \
make -C /lib/modules/2.6.12-gentoo-r10/build 
 SUBDIRS=/usr/local/src/zaptel-1.2.3 INSTALL_MOD_PATH= 
 INSTALL_MOD_DIR=misc modules_install; \
if ! [ -f wcfxsusb.ko ]; then \
rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxsusb.ko; \
fi; \
rm -f

Re: [Asterisk-Users] asterisk logger - urgent!!!

On Thu, Feb 09, 2006 at 11:56:02AM -0200, Dov Bigio wrote:
 Hi Tzafrir,
 
 The problem was the file Master.csv that had reached 2.0GB.
 I am writing a cron script to backup this file periodically and prevent this
 from happening.
 
 Any way, if any developers are reading this, I don't think that rotating
 asterisk logs is the best way to handle this problem!
 Maybe a more user-friendly message could be logged, infoming which file
 reached the 2.0GB.

This should be the job of the log rotate script. Limit its size to 1 GB
and rotate it weekly. see logrotate.conf(5) . Part of just about any linux 
distro. Normally run in the daily cron.

 
 About your question, yes I do, for log files. Is logger rotate could also
 after I delete csv files?

Anybody?

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

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[Asterisk-Users] Meetme echo cancellation

2006-02-09 Thread Steven Langley

Title: Meetme echo cancellation






Hi there

I am using IAX2 softphones dialing into a meetme conference. In my softphone I was forcing uses to click on a button when they wanted to speak, enabling their microphone and disabling their speakers. This way when a user was speaking they did not hear their voice half a second later (because meetme mixes the voice and sends to everyone in the conference).

Now because of requirements there is a need for users not to have to click a button when speaking (and have their microphones and speakers enabled at all times)  much like Skype. How would I prevent a user hearing their own voice half a second later? Using some kind of echo cancellation? I am not sure that this is defined as echo though.

Does anyone have any ideas?

Many thanks

Steven


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Re: [Asterisk-Users] ztdummy on gentoo 2005.1

2006-02-09 Thread Gerard Saraber

Invalid module format usually means that the module you're trying to
load was not compiled with the same parameters as the kernel you're
trying to load it into, make sure your /usr/src/linux symlink points to
the kernel you are actually running, /var/log/messages etc. will usually
have more specific information on why the module format is 'wrong' .

I would suggest after checking the /usr/src/linux symlink, to recompile
the kernel, the ztdummy module and booting into the newly compiled
kernel. its possible that all it takes is to recompile the module
though.

Regards,
Gerard Saraber
[EMAIL PROTECTED]

On Thu, 2006-02-09 at 09:30 -0600, Miguel wrote:
 I changed the line with your sugestion but same result (after reboot):
 
 voip # lsmod
 Module  Size  Used by
 voip # modprobe ztdummy
 FATAL: Error inserting ztdummy 
 (/lib/modules/2.6.12-gentoo-r10/misc/ztdummy.o): Invalid module format
 FATAL: Error running install command for ztdummy
 voip mmiranda #
 
 ---
 Miguel
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-- 
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]

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[Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring?

2006-02-09 Thread Andrew Kohlsmith

This is my first foray into SIP telephony, so be gentle.  :-)

The Polycom SoundPoint IP 501 phones have been fantastic so far.  I still have 
a lot to learn when it comes to them, but the manual seems pretty extensive 
and so far Asterisk has been playing well with them.

I have a need to be able to identify incoming calls based on some factor 
(could be time of day, caller ID, dialed number, it doesn't matter.) -- 
Assuming Asterisk can differentiate between the calls I want, how do I inform 
the IP501?  There are only three line appearances -- I can't simply just 
ring a different appearance since there aren't enough of them.

Is there a way to get Asterisk to tell the IP501 to use a different ring, put 
something up on the display, *something* on a dynamic basis?  The wiki 
doesn't seem to have a lot of information about this kind of thing.

example:  I am using the first two appearances for shared lines, and the third 
for my own two-way extension.  I'd like the phone to be able to tell me 
(different ring, something on the display, both?) that an incoming call was 
for customer service (ring ring), sales (woop woop), my wife 
(ah-wooh-gah), a non-business-related contract caller (beep beep), that 
my laundry is done (buzz), you name it..  Assuming I have the dialplan 
correctly differentiating between these types of calls, how do I get the 
phone to notify me in different ways?

-A.
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Re: [Asterisk-Users] TDM400p

Hans Witvliet wrote:

 Does the TDM400 not only fits, but also functions in a 3.3V only slot?
 
From what i detected so far, is that some MOBO manufactures have
 pci-slots that provide 3.3 Volt AND 5.0 Volt, thus can handle all kind
 of cards.

The TDM400P and TDM2400P will work in any PCI or PCI-X slot, as long as
the motherboard implements at least PCI 2.2.
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Re: [Asterisk-Users] asterisk logger - urgent!!!

Dov Bigio wrote:

 Any way, if any developers are reading this, I don't think that rotating
 asterisk logs is the best way to handle this problem!
 Maybe a more user-friendly message could be logged, infoming which file
 reached the 2.0GB.

Unfortunately when we receive SIGFSZ from the kernel, we have no way to
know which file caused it. The assumption in Asterisk is that the only
files we write to that will ever reach that size are log files. If any
other file does, there will be trouble, as you have seen.
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[Asterisk-Users] tdm400p setup in china question

2006-02-09 Thread Cavanna, Richard

I am thinking of setting up a * system for a remote office in china.  I
was going to use a tdm400p to setup a basic 3X8 system.

I will setup the system in the US and ship it over.

Does anyone know of any problems that I should watch out for.
Signaling, caller id, ..
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Re: [Asterisk-Users] Dell PowerEdge 1800 and TE410P

Andres wrote:

 We run the TE410 on the PowerEdge 1850 and it is rock solid.  We do not
 have problems with the onboard ethernet controller.

The interaction between the TE cards and the onboard ethernet controller
affects only a small number of users, and we don't know what
specifically causes it (kernel driver versions, BIOS versions, etc).

In general, the PEx8x0 machines are fine with Digium's interface cards.
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Re: [Asterisk-Users] more cpu intensive echo cancellers ?

2006-02-09 Thread Matthew Fredrickson



On Feb 8, 2006, at 1:03 PM, Gerard Saraber wrote:


Hi,
I've had some decent luck with the mark3 echo canceller from the zaptel
driver, echos on about 20% of the calls, people I've called say I sound
great now, but our side hears echos.
I was wondering if there was any way to tweak the current software
cancelers into using more CPU (and hopefully doing a better job, close
to a hardware canceler), I only have 10 lines, and a single call takes
0.5% cpu, I would have no problem if it went up to 5-10% if they would
work better.
Or should I just give up now and buy the channel bank, tellabs hardware
echo canceler and a T1 pci card? (hope TDM400P cards have decent resale
value ;)



Yeah there is, upgrade to trunk and use the new echo canceller there 
(MG2).  It's supposed to rock, at least from what I've heard.  All the 
MEC cancellers are _OLD_.  At least switch to 1.2 and the KB1 echo 
canceler before giving up.


---
Matthew Fredrickson

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Re: [Asterisk-Users] stable ISDN BRI card for asterisk

2006-02-09 Thread Armin Schindler

On Thu, 9 Feb 2006, Peng Yong wrote:
 we would like to running 4 port or 8 port ISDN BRI card on production asterisk
 system.
 
 any one can recommend a good product? is it stable and with good voice
 quality?

I have very good results with Eicon DIVA Server 4BRI
 http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm
or
 http://www.eicon.com/worldwide/products/MediaGateways/diva-server-v4bri.htm

Armin
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Re: [Asterisk-Users] Meetme echo cancellation

Steven Langley wrote:

 I am using IAX2 softphones dialing into a meetme conference. In my softphone
 I was forcing uses to click on a button when they wanted to speak, enabling
 their microphone and disabling their speakers. This way when a user was
 speaking they did not hear their voice half a second later (because meetme
 mixes the voice and sends to everyone in the conference).

This is wrong. app_meetme does not send the speaker's own voice back to
them, because the mixing in Zaptel removes it for the speaker's channel
in the conference.

Try it :-)
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[Asterisk-Users] I need help on VICIDIAL and auto dial

2006-02-09 Thread Vic Jolin

Vicidial can't call and transfer to my softphone. 

I get some line that says 

Spawn Extensionexited on non zero

Here's some of the CLI output. I am using Asterisk 1.2.4 and astguiclient 1.1.8


...thanks for the help



|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|
-- VDAD get agent: |0|update of vla table: |127.0.0.1|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='
127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|

|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|-- VDAD get agent: |0|update of vla table: |127.0.0.1
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|


|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|-- VDAD get agent: |0|update of vla table: |127.0.0.1
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|


|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|-- VDAD get agent: |0|update of vla table: |127.0.0.1
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|


|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|-- VDAD get agent: |0|update of vla table: |127.0.0.1
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|


|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|-- VDAD get agent: |0|update of vla table: |127.0.0.1
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|
 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 -- Hungup 'Zap/pseudo-1749551349' == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/800-be53'
 -- Executing DeadAGI(SIP/800-be53, call_log.agi|h) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi

|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|-- VDAD get agent: |0|update of vla table: |127.0.0.1
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|
+ CALL LOG HUNGUP: |1139414568.0|SIP/800-be53|h|2006-02-09 0:03:58|min: | -- AGI Script call_log.agi completed, returning 0

|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|-- VDAD get agent: |0|update of vla table: |127.0.0.1
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|


|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time   and lead_id != '';|-- VDAD get agent: |0|update of vla table: |127.0.0.1
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid='1139414618.5', channel='IAX2/u32218094-6', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time  '19700101075955' limit 1;|
 == Manager 'sendcron' logged off from 127.0.0.1
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Re: [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring?

Andrew Kohlsmith wrote:

 Is there a way to get Asterisk to tell the IP501 to use a different ring, put 
 something up on the display, *something* on a dynamic basis?  The wiki 
 doesn't seem to have a lot of information about this kind of thing.

There are examples (IIRC) of making the phone auto-answer for specific
types of calls; those should get you started, since they demonstrate how
to have the phone choose a different 'alerting' configuration on a
call-by-call basis.
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[Asterisk-Users] Issues in Australia? Ringing, iaxy etc

2006-02-09 Thread Chris Earle \(CBL\)

Hi all,

getting a server going wiht a few TDM400's and some phones, and some IAXys
too

I haven't heard any issues about AU phones being able to RING in Australia,
like the problem in the UK with ring capacitors on the BT system.  Are there
any problems like that?

Also, with the iaxy's -- they should work (and ring) in Australia right?
The only hint I'm seeing around is the use of notransfer=yes in the iax.conf
for the iaxy entry

Basically, just hoping for a smooth transition over to the asterisk
system

Cheers


--
Chris Earle
System Solutions Specialist


-- 
This message has been scanned for viruses and dangerous content by
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Re: [Asterisk-Users] ztdummy on gentoo 2005.1 [SOLVED]

2006-02-09 Thread Miguel


Tzafrir Cohen wrote:


Makefile:204: target `ztdummy.o' given more than once in the same rule.
   



Something is bad. Did you edit Makefile?
 

 


Yes. I delete all the modules, leaving the lines like this

before:

MODULES:=zaptel tor2 torisa wcusb wcfxo wctdm wctdm24xxp \
ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp pciradio \
ztd-loc # ztdummy


after:

MODULES:= ztdummy

but some lines below you find this:

# build ztdummy by default for 2.6 kernels
ifeq (${BUILDVER},linux26)
MODULES+=ztdummy
endif

So, the ztdummy is always compiled with 2.6 kernels, i changed the line 
commenting out all modules :



MODULES:=#zaptel tor2 torisa wcusb wcfxo wctdm wctdm24xxp \
ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp pciradio \
ztd-loc # ztdummy
#MODULES+=wcfxsusb
# build ztdummy by default for 2.6 kernels
ifeq (${BUILDVER},linux26)
MODULES+=ztdummy
endif


and edited the /etc/modules.d/zaptel file leaving only:

voip zaptel-1.2.3 # cat /etc/modules.d/zaptel
post-install ztdummy /sbin/ztcfg


now when i load the module without problems:

voip # lsmod
Module  Size  Used by
voip # modprobe ztdummy
voip # lsmod
Module  Size  Used by
ztdummy 2468  -
zaptel44  -


I notice that the zaptel module is loaded too, is this normal?
---

   Miguel



Something is wrong. Not sure exactly what.

Now the overly-complicated install target's script springs into action
and manages to install the .o file into the modules dir. Funny, don't
you think?

 


CC  /usr/local/src/zaptel-1.2.3/ztdummy.mod.o
LD [M]  /usr/local/src/zaptel-1.2.3/ztdummy.ko
make[1]: Leaving directory `/usr/src/linux-2.6.12-gentoo-r10'
cc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so 
zonedata.lo tonezone.lo

install -m 444 udev/zaptel.rules-combined /etc/udev/rules.d/zaptel.rules
install -D -m 755 ztcfg /sbin/ztcfg
if [ -f sethdlc-new ]; then \
  install -D -m 755 sethdlc-new /sbin/sethdlc; \
elif [ -f sethdlc ]; then \
  install -D -m 755 sethdlc /sbin/sethdlc ; \
fi
if [ -f zttool ]; then install -D -m 755 zttool /sbin/zttool; fi
if [ -f zaptel.ko ]; then \
  for x in ztdummy.ko ztdummy.ko; do \
  rm -f /lib/modules/2.6.12-gentoo-r10/extra/$x ; \
  done; \
  make -C /lib/modules/2.6.12-gentoo-r10/build 
SUBDIRS=/usr/local/src/zaptel-1.2.3 INSTALL_MOD_PATH= 
INSTALL_MOD_DIR=misc modules_install; \

  if ! [ -f wcfxsusb.ko ]; then \
  rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxsusb.ko; \
  fi; \
  rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxs.ko; \
else \
  for x in ztdummy.o ztdummy.o; do \
  install -D -m 644 $x /lib/modules/2.6.12-gentoo-r10/misc/$x ; \
  done; \
  if ! [ -f wcfxsusb.o ]; then \
  rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxsusb.o; \
  fi; \
  rm -f /lib/modules/2.6.12-gentoo-r10/misc/wcfxs.o; \
fi
install -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0
[ `id -u` = 0 ]  /sbin/ldconfig || :
rm -f /usr/lib/libtonezone.so
ln -sf libtonezone.so.1.0 \
  /usr/lib/libtonezone.so.1
ln -sf libtonezone.so.1.0 \
  /usr/lib/libtonezone.so
if [ -x /usr/sbin/sestatus ]  (/usr/sbin/sestatus | grep SELinux 
status: | grep -q enabled) ; then restorecon -v 
/usr/lib/libtonezone.so; fi

install -D -m 644 zaptel.h /usr/include/linux/zaptel.h
install -D -m 644 torisa.h /usr/include/linux/torisa.h
install -D -m 644 tonezone.h /usr/include/tonezone.h
install -m 644 doc/ztcfg.8 /usr/share/man/man8
install -m 644 doc/zttool.8 /usr/share/man/man8
if [ -n /etc/modules.d/zaptel ]; then \
  if [ -f /etc/modules.d/zaptel ]; then mv -f /etc/modules.d/zaptel 
/etc/modules.d/zaptel.bak ; fi; \

  cat /etc/modules.d/zaptel.bak | grep -v alias char-major-250 | \
  grep -v post-install torisa /sbin/ztcfg | \
  grep -v post-install wcfxsusb /sbin/ztcfg | \
  grep -v alias wctdm | \
  grep -v post-install wctdm /sbin/ztcfg  /etc/modules.d/zaptel; \
  if ! grep options torisa /etc/modules.d/zaptel; then \
  echo options torisa base=0xd  /etc/modules.d/zaptel; \
  fi; \
  if ! grep alias char-major-196 /etc/modules.d/zaptel; then \
  echo alias char-major-196 torisa  /etc/modules.d/zaptel; \
  fi; \
  for x in ztdummy ztdummy; do \
  if ! grep -q post-install $x /etc/modules.d/zaptel; then \
  if ! grep -q install $x  /etc/modules.d/zaptel; then \
  if [ $x != zaptel ] ; then \
  if [ -f zaptel.ko ]; then echo install $x 
/sbin/modprobe --ignore-install $x  /sbin/ztcfg  
/etc/modules.d/zaptel; \
  else echo post-install $x /sbin/ztcfg  
/etc/modules.d/zaptel; \

  fi; \
  fi; \
  fi; \
  fi; \
  done; \
  if ! grep ias wcfxs /etc/modules.d/zaptel; then \
  echo alias wcfxs wctdm  /etc/modules.d/zaptel; \
  fi; \
  if ! grep alias wct2xxp /etc/modules.d/zaptel; then \
  echo alias wct2xxp wct4xxp  /etc/modules.d/zaptel; \
  fi; \
fi
options torisa base=0xd
alias char-major-196 torisa
if [ -d /etc/modutils ]; then \

RE: [Asterisk-Users] sipura 3000 and other probs

2006-02-09 Thread john

With pen in hand, Technical Support succussfully stormed bulwarks which
others armed with sword and excommunication have been repulsed, and said
...
 There's a vox forum that focuses on Sipuras - post your query there for
 good tech help.  We've deployed a number of Sipura's and haven't
 experienced that problem (yet).  Have you started with the basics:
 firmware version, analog cabling, etc.

 MD


Oh yeah... ex Satcom/ISDN tech here and so I checked all the basics,
swapped caples, firmware is the latest, etc. I'll check on the voxilla
forum, that completely slipped my mind.

I was hoping that maybe someone on the list had seen something like this,
particularly given the error from the log I described below:

chan_sip.c: That's odd...  Got a response on a call we dont know about.
Cseq 102 Cmd SIP/2.0

Does anyone have the time to let me know exactly what this error points
to? When I say exactly, of course I don't mean exactly in my configuration
which none of you have seen, I mean, what is the software seeing that it
isn't prepared to handle. As I also mentioned, I looked at the code, but
I'm an amateur programmer at best, with little experience, not to mention
I'm still learning about the SIP protocol, so I'm just not sure what this
is telling me.

I strongly suspect that this is probably tied in with my situation, but
according to all the docs/forums/setups I've researched, my setup looks
OK.

Regards,

John C.

 -Original Message-
 From: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] sipura 3000 and other probs

...
...
 For example, I'll be talking to an incoming caller and an echo starts
 quietly in the background. Within a minute or so, I'll lose the
 connection,
 with my voice 100% reflecting back at me, and the caller says the same
 occurs at their end, his/her voice 100% reflecting back to him/her.

 I usually have to reset the device completely at that point. Other times,
 I'll get a call and talk for 1/2 hour with no problems whatsoever.

 I have it set up with [EMAIL PROTECTED] V 2.2 according to the setup at

   http://mundy.org/blog/index.php?p=65

 Also I get the following strange error:

 chan_sip.c: That's odd...  Got a response on a call we dont know about.
 Cseq 102 Cmd SIP/2.0

 I can't seem to pin it down. I've checked the source (chan_sip.c) and
 because I'm not well aquainted with the protocol, I'm about as clueless
 as could be as to what its telling me.
...


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RE: [Asterisk-Users] What ATA should I buy?

2006-02-09 Thread Sam Tam

We have got some ATA for only $55 if you are interested?

Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Sampson
Sent: Thursday, February 09, 2006 11:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] What ATA should I buy?

I've used the spa-1001 and the spa-2001 for faxes. Works good over a 
local area network. thevoipconnection sells those for about 60 bucks though.

Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

Tomislav Parčina wrote:

I have running * without any Digium (or any other) hardware. Now I need to
connect analog FAX machine to it. I think that cheapest and easiest way is
to buy ATA. Please correct me if I'm wrong.

Now, which ATA should I buy? Local dealer sells those four. I can buy
something else (if there is any reason for it), but I prefer something of
this. 

One more question, can I plug two lines in any of those ATA-s?

Sipura SPA-2100 SIP-ATA160$
Sipura SPA-1001 SIP-ATA125$
ALL7902 IP SIP ATA Adapter / Router106$
Grandstream HandyTone ATA486   142$

Thank you for any suggestions.

P.S.
If this is second time you see this message, then sorry for resending, but
I didn't see it on list...

--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
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[Asterisk-Users] IP Authorization

2006-02-09 Thread Sam Tam











I think this is a
question that has been discussed before. 
But you see nowadays most carriers will provide thing like SIP using IP
authorization rather than username and password and I am now wondering whether
Asterisk can do something like that or not?



Sam








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Re: [Asterisk-Users] asterisk logger - urgent!!!

2006-02-09 Thread Bob Goddard

On Thursday 09 Feb 2006 16:01, Kevin P. Fleming wrote:
 Dov Bigio wrote:
  Any way, if any developers are reading this, I don't think that rotating
  asterisk logs is the best way to handle this problem!
  Maybe a more user-friendly message could be logged, infoming which file
  reached the 2.0GB.

 Unfortunately when we receive SIGFSZ from the kernel, we have no way to
 know which file caused it. The assumption in Asterisk is that the only
 files we write to that will ever reach that size are log files. If any
 other file does, there will be trouble, as you have seen.

Why use fputs which give you no indication of the type of error
when the raw write does?


B

-- 
http://www.mailtrap.org.uk/
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Re: [Asterisk-Users] more cpu intensive echo cancellers ?

2006-02-09 Thread Gerard Saraber

On Thu, 2006-02-09 at 10:05 -0600, Matthew Fredrickson wrote:
 Yeah there is, upgrade to trunk and use the new echo canceller there 
 (MG2).  It's supposed to rock, at least from what I've heard.  All the 
 MEC cancellers are _OLD_.  At least switch to 1.2 and the KB1 echo 
 canceler before giving up.
 
 ---
 Matthew Fredrickson
 

Aha! good to know, I am running asterisk 1.2.4 and zaptel-1.2.3, should
I switch to CVS ? I've tried the MG2 canceler with the above versions,
each time I tried it, I had a constant echo, where with the mark3 it
went away after a second or two at the beginning of the call. (which I
can live with, but some of the calls are completely unusable due to
continuous or returning echos)
I'll go play with the mg2 and kb1 again and see what happens

-- 
Thanks,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]

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[Asterisk-Users] ztmonitor output weirdness

2006-02-09 Thread Ronald Hartmann

Good Day,

I have a weird issue using zaptel-1.2.3 and a PRI with 8 voice
channels.

With nobody on the phone using ztmonitor I get the following:


Why would I have such high TX signals on certain channels.

~ron

[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 1 -vv

Visual Audio Levels.

 Use zapata.conf file to adjust the gains if needed.

( # = Audio Level  * = Max Audio Hit )
(RX)
(TX)
##*
Rx: 8 (8) Tx:  2628 ( 2628)
[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 2 -vv

Visual Audio Levels.

 Use zapata.conf file to adjust the gains if needed.

( # = Audio Level  * = Max Audio Hit )
(RX)
(TX)
 
Rx: 0 (0) Tx:22 (   22)
[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 3 -vv

Visual Audio Levels.

 Use zapata.conf file to adjust the gains if needed.

( # = Audio Level  * = Max Audio Hit )
(RX)
(TX)
*
Rx:   130 (  130) Tx:  1637 ( 1637)
[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 4 -vv

Visual Audio Levels.

 Use zapata.conf file to adjust the gains if needed.

( # = Audio Level  * = Max Audio Hit )
(RX)
(TX)
 
#* Rx: 2 (2) Tx:  6140 (
6140)
[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 5 -vv

Visual Audio Levels.

 Use zapata.conf file to adjust the gains if needed.

( # = Audio Level  * = Max Audio Hit )
(RX)
(TX)
 
Rx: 0 (0) Tx: 0 (0)
[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 6 -vv

Visual Audio Levels.

 Use zapata.conf file to adjust the gains if needed.

( # = Audio Level  * = Max Audio Hit )
(RX)
(TX)
 
Rx: 0 (0) Tx: 0 (0)
[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 7 -vv

Visual Audio Levels.

 Use zapata.conf file to adjust the gains if needed.

( # = Audio Level  * = Max Audio Hit )
(RX)
(TX)
 
Rx: 0 (0) Tx: 0 (0)
[EMAIL PROTECTED] zaptel-1.2.3]# ./ztmonitor 8 -vv

Visual Audio Levels.

 Use zapata.conf file to adjust the gains if needed.

( # = Audio Level  * = Max Audio Hit )
(RX)
(TX)
 
Rx: 0 (0) Tx: 0 (0)

Ronald Hartmann
Director Technical Services
VerCom Systems, Inc.
410 Fame Rd, Dayton, OH 45449 
Voice:866.VerCom.4 Fax: 866.422.6486


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Re: [Asterisk-Users] asterisk logger - urgent!!!