date:20041213

 I wonder what does this warning 399 mean and how to workaround?  sip show
 peers says that sip client is unreachable althought it works with some
 eexceptions ...

http://www.google.com/search?q=%22detected+NAT+type+is+full+cone%22
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[Asterisk-Users] music on hold garbled

2004-12-13 Thread Jay Austad

Anyone have an easy fix for making my music on hold to work properly?  
It's very loud and has a lot of garbling in it.  X is not running, and 
the framebuffer is disabled.

I've tried just about every example I could find.  I just uploaded 
standard mp3's, but even the ones that came with it sounded the same.

~jay
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Re: [Asterisk-Users] music on hold garbled

 It's very loud and has a lot of garbling in it.  

What/how many phones have you tried it on? What channels
(ZAP/SIP/IAX2) and what codecs?
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Re: [Asterisk-Users] Re: Cant set H323 up

2004-12-13 Thread Rodolfo Grave

Sorry... Im running SuSE 9.0
kido noagbodji wrote:
what os are you running?
K.
- Original Message - 
From: Rodolfo Grave [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion [EMAIL PROTECTED]
Sent: Monday, December 13, 2004 1:27 AM
Subject: Re: [Asterisk-Users] Re: Cant set H323 up

Hi
Now I do have compiled all the libraries, and added the
load = chan_h323.so
in the modules.conf file. Actually, now asterisk is attempting to load
the chan_h323.so module. The problem is that Im getting this error now:
 [chan_h323.so]Dec 13 02:24:01 WARNING[12023]: loader.c:258
ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object
file: No such file or directory
I've moved the libpt_linux_x86_r.so.1.5.2 file to /usr/lib,
/usr/lib/asterisk, /usr/lib/asterisk/modules
After each move, I ran ldconfig the error was always the same...
does anyone know where does asterisk looks for this file? Or if the
cause for this is another?
Im using the H323 channel included in the Asterisk tree.
Thanks,
RODOLFO
Corvin wrote:
Rafael J. Risco G.V. wrote:

On Sat, 11 Dec 2004 16:49:12 +, Corvin [EMAIL PROTECTED] wrote:

Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa:

Hi.
I need to set up H323 on an Asterisk box. I've succesfuly compiled the
asterisk oh323 (including of course all the dependencies: PWlib and
OpenH323), and then compiled asterisk. However, asterisk doesn't
report
a registered H323 channel (when it starts, it reports IAX2, ZAP and
SIP
channels, however, the 323 word doesn't appear in the whole output).
Is there anything I'm missing? I've read the documentation on the
wiki,
and none said nothing about editing a config file. I did noticed that
they talked about the oh323.conf file, which I dont have.

BTW. you should check direcory with oh323 a there should be
asterisk-driver
directory and there you find sample config.
Then you sghould load module in modules.conf.
BTW. I can't still compile any h323 driver :(((.
Corvin
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[Asterisk-Users] outgoing call queue.

2004-12-13 Thread Pedro N.

Hi all,
is it possible to make a queue for outgoing calls? That's for preventing
Device '/dev/ttyI 0' is busy error when having only one line to dialout
and many files in /var/spool/asterisk/outgoing folder. So it would call
only one call at the time and when it's done it would move to next.

Thanx in advance.

~pete

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[Asterisk-Users] Strange Segmentation fault

2004-12-13 Thread Paradise Dove

I get seg. fault with my * box. at the crash time i had about 35
Bridged Channel.
i have:
- dual xeon box (3.2Ghz)
- 2Gb of memory 
- E7501 chipset motherboard.
- U320 scsi disks
- intel Gb ethernet device.
- i only use sip for clients (no fxs in box)
- TE405P for fxo (with 4 atran TA750).
- ulaw is used as codec and echo cancellationo is enabled.

but the core dump file has nothing to show with gdb.

this is the output of gdb:

Program terminated with signal 11, Segmentation fault.
#0  0xb7fbbce4 in ?? ()
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[Asterisk-Users] IAX.cc / Sixtel?

2004-12-13 Thread Me

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad.
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
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[Asterisk-Users] Traditional Telephony Interface Card

2004-12-13 Thread Stojan Sljivic - Pamet

Title: Message



Hi 
all,

We are 
located in Europe and we have four analog telephony lines.
What 
hardware is needed to connect Asterisk with these lines?
WhatVoIP hard phones operate best with 
Asterisk?

Regards,
Stojan 
Sljivic
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Re: [Asterisk-Users] Traditional Telephony Interface Card

2004-12-13 Thread Michael Bielicki

A tdm40B 4 FXS card from digium. WE can deliver that to you, we are
even setting up a reseller in Belgrade. Please contact me off list for
details


On Mon, 13 Dec 2004 12:46:49 +0100, Stojan Sljivic - Pamet
[EMAIL PROTECTED] wrote:
  
 Hi all, 
   
 We are located in Europe and we have four analog telephony lines. 
 What hardware is needed to connect Asterisk with these lines? 
 What VoIP hard phones operate best with Asterisk? 
   
 Regards, 
 Stojan Sljivic 
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-- 
Michal Bielicki
http://www.asterisk.com.pl/
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[Asterisk-Users] Echo on one E1 line, but not the other

2004-12-13 Thread Asterisk

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p
EuroISDN.

We have 2 ISDN lines, one we had for testing, and one for general (40+
users) use.

During the testing phase, we had 10 phones linked to the second ISDN line,
and there were no problems with echo at all. Lucky me. However, since we
have started rolling out, we've had quite loud complaints that there is a
terrible echo. If I direct my 7960 to use the primary line, there is an
echo. If I use the second line (dialling the same external number) there is
no echo at all.

What could be the issue ? I have noticed on the primary line there is a
detected rx/tx on channel xx, echo cancellation disabled (or something
like that. Is this the cause ? We have several fax / modems going through
the line - should I always dedicate a channel to them ?

I've included my zaptel.conf and zapata.conf file below. Any help / comments
appreciated.

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

span=3,2,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78

span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109

loadzone=uk
defaultzone=uk

#

;zapata.conf

[general]

[trunkgroups]

[channels]
musiconhold=default
language=en
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callerid=asreceived
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
;callgroup=1
;pickupgroup=1
;adsi=yes

context=isdn32-b
pridialplan=unknown
group=1
signalling=pri_cpe
switchtype=euroisdn
channel=1-15,17-31

context=meridian-b
group=2
signalling=pri_net
switchtype=euroisdn
channel=32-46,48-62

context=isdn32-a
pridialplan=unknown
group=3
signalling=pri_cpe
switchtype=euroisdn
channel=63-77,79-93

context=meridian-a
group=4
signalling=pri_net
switchtype=euroisdn
channel=94-108,110-124

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[Asterisk-Users] transferring variables with IAX2?

2004-12-13 Thread Roy Sigurd Karlsbakk

hi
is it, or can it be possible to transfer stuff like HANGUPCAUSE or 
RDNIS over IAX2? This is really a nessicity for multi-server setups to 
become any good...

roy
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Re: [Asterisk-Users] Traditional Telephony Interface Card

2004-12-13 Thread Andy Burns

Michael Bielicki wrote:
 Stojan Sljivic wrote:

 We are located in Europe and we have four analog telephony lines.
 What hardware is needed to connect Asterisk with these lines?

 A tdm40B 4 FXS card from digium. WE can deliver that to you, we are
 even setting up a reseller in Belgrade.
Hopefully if you are setting up as a reseller you'll learn the diffence 
between an FXS and an FXO, in this case a TDM04B not a TDM40B!

P.S. Bottom posting would be nice too ;-)
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Re: [Asterisk-Users] transferring variables with IAX2?

2004-12-13 Thread Peter Svensson

On Mon, 13 Dec 2004, Roy Sigurd Karlsbakk wrote:

 is it, or can it be possible to transfer stuff like HANGUPCAUSE or 
 RDNIS over IAX2? This is really a nessicity for multi-server setups to 
 become any good...

There is a patch floating around (on the mailing list and/or on the bug 
tracker) that transports the HANGUPCAUSE over IAX2 in a text message. 
Perhaps this could be generalized to allow any user defined variable to be 
passed?

Peter


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RE: [Asterisk-Users] Echo on one E1 line, but not the other

2004-12-13 Thread Steve Hanselman

It looks like this is a splice between a couple of ISDN-30 lines and one or
more PBX's?

Are they both with the same provider, or with different providers?

We ended up adjusting the gain our ours as we would hear a distinct echo on
certain calls.

Other than that, you'll need to do the usual tests, check for shared
interrupts, also, see if disk activity causes a problem.

(check /proc/interrupts for shared interrupts).

Have you also checked the output of the pri commands to ensure that you're
not getting line errors?

Steve


-Original Message-
From: Asterisk [mailto:[EMAIL PROTECTED] 
Sent: 13 December 2004 12:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Echo on one E1 line, but not the other

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p
EuroISDN.

We have 2 ISDN lines, one we had for testing, and one for general (40+
users) use.

During the testing phase, we had 10 phones linked to the second ISDN line,
and there were no problems with echo at all. Lucky me. However, since we
have started rolling out, we've had quite loud complaints that there is a
terrible echo. If I direct my 7960 to use the primary line, there is an
echo. If I use the second line (dialling the same external number) there is
no echo at all.

What could be the issue ? I have noticed on the primary line there is a
detected rx/tx on channel xx, echo cancellation disabled (or something
like that. Is this the cause ? We have several fax / modems going through
the line - should I always dedicate a channel to them ?

I've included my zaptel.conf and zapata.conf file below. Any help / comments
appreciated.

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

span=3,2,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78

span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109

loadzone=uk
defaultzone=uk

#

;zapata.conf

[general]

[trunkgroups]

[channels]
musiconhold=default
language=en
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callerid=asreceived
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
;callgroup=1
;pickupgroup=1
;adsi=yes

context=isdn32-b
pridialplan=unknown
group=1
signalling=pri_cpe
switchtype=euroisdn
channel=1-15,17-31

context=meridian-b
group=2
signalling=pri_net
switchtype=euroisdn
channel=32-46,48-62

context=isdn32-a
pridialplan=unknown
group=3
signalling=pri_cpe
switchtype=euroisdn
channel=63-77,79-93

context=meridian-a
group=4
signalling=pri_net
switchtype=euroisdn
channel=94-108,110-124

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[Asterisk-Users] Asterisk receiving SER calls

2004-12-13 Thread Joao Pereira





Hi
Im trying to make Asterisk receiev SER calls and 
then redirect them to GNUGK.
But until now, Asterisk isnt receiving 
nothing...


Asterisk is already as a gateway in GNUGK as shown in the gnugk 
monitorization:
RCF|Asterisk_ip:Asterisk_port|asterisk:h323_ID=ASTERISK:h323_ID=664:dialedDigits|gateway|8478_endp

I installed oh323 for Asterisk. The versions I have are the 
following:
pwlib-1.6.6-0_11.rh9openh323-1.13.5-0_13.rh9gnugk-2.0.8-linux-x86
.






In ser.cfg I have just this:
rewritehostport("Asterisk_ip : 
Asterisk_port"); 
t_relay();

that should be enough.




in 
sip.conf I have: 
[general]context=defaultautocreatepeer=yescanreinvite=no 
[mic-inout]type=friendsecret=**username=asterisk
fromuser=asteriskhost=my.domain.com.pedtmfmode=rfc2833insecure=very

And in extensions.conf I putted
exten = 
_00NXX.,1,Dial,OH323/${EXTEN}


What is missing to make Asterisk receive calls from 
SER?

Joao Pereira
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Re: [Asterisk-Users] transferring variables with IAX2?

2004-12-13 Thread Roy Sigurd Karlsbakk

is it, or can it be possible to transfer stuff like HANGUPCAUSE or
RDNIS over IAX2? This is really a nessicity for multi-server setups to
become any good...
There is a patch floating around (on the mailing list and/or on the bug
tracker) that transports the HANGUPCAUSE over IAX2 in a text message.
Perhaps this could be generalized to allow any user defined variable 
to be
passed?
I beleive I read some discussion on the topic, and compared to the php 
register_globals case. we'll probably need some way to distinguish an 
external variable (sent via IAX2) and an internal variable (global or 
not).

roy
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RE: [Asterisk-Users] Strange Segmentation fault

2004-12-13 Thread Florian Overkamp

Hi, 

 -Original Message-
 I get seg. fault with my * box. at the crash time i had about 35
 Bridged Channel.
 i have:
 - dual xeon box (3.2Ghz)
 - 2Gb of memory 
 - E7501 chipset motherboard.
 - U320 scsi disks
 - intel Gb ethernet device.
 - i only use sip for clients (no fxs in box)
 - TE405P for fxo (with 4 atran TA750).
 - ulaw is used as codec and echo cancellationo is enabled.
 
 but the core dump file has nothing to show with gdb.
 
 this is the output of gdb:
 
 Program terminated with signal 11, Segmentation fault.
 #0  0xb7fbbce4 in ?? ()

Hmm without knowing anything else about your specific situation: A signal 11
most often is caused by a hardware malfunction, for instance a rotten bit in
your memory or something.Any chance you could do some heavy diagnostics on
that machine ?

http://www.bitwizard.nl/sig11/

Best regards,
Florian

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Re: [Asterisk-Users] Re: Recommendations for full featured phones

2004-12-13 Thread Carmi Weinzweig

Are you replacing a Merlin Legend (hybrid PBX/key system) or a Merlin 
4/10, 8/20 low end key system? You should be aware that in its current 
form, Asterisk does not support shared extensions something commonly 
used in most key environments.

/carmi
On Dec 6, 2004, at 9:37 AM, Pavel Jezek wrote:
look at:
http://netphone.intracom.gr/english.htm
we have order this meanwhile for lab testing,
so I would be able to refer for about a month...
PJ
- Original Message -
From: Sean Cook
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Monday, December 06, 2004 12:19 AM
Subject: Recommendations for full featured phones
We are considering a replacement of a legacy PBX system (merlin).  I am
trying to figure out which phones would be best supported with the
fullest set of features.  Any recommendations?
Sean
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[Asterisk-Users] Dialing out to 2 clients simultaneously

2004-12-13 Thread niels


Hi

When I register a SIP or IAX client to asterisk and I dial to it from
another UA then there is no problem at all 

But, when I register two or more clients to the SAME peer (with the same
user/pass) and I call to this peer.. Then only the UA which registered
the last will ring.. Others don't ring... 

What can I do about this?? 

I would like to register for example 10 UA's to the same peer and want
them all to ring at the same time without having to set up different
usernames and passwords for all these ua's and having to make difficult
dialplans

Is this possible? Am I doing something wrong or is this behaviour by
design?

Regards,
Niels

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Re: [Asterisk-Users] What route do diverted SIP calls travel?

2004-12-13 Thread Sam Bashton

On Mon, 13 Dec 2004 11:40:46 +, Andy Burns
[EMAIL PROTECTED] wrote:
 If I have inbound SIP calls arriving from a provider's gateway to an
 asterisk server on my LAN, which then routes the call back out via the
 provider's gateway to a PSTN number, once the call is answered do all
 the voice packets pass through my asterisk PBX, or is SIP intelligent
 enough to patch the two PSTN ends of the call direct to each other going
 only via two ports on the provider's gateway?

The data-heavy portion of the traffic is RTP, and that should be a
direct connection using your providers gateway.  Make sure you have
'canreinvite=yes' set in the appropriate section of your sip.conf.

-- 
Sam Bashton
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Re: [Asterisk-Users] Traditional Telephony Interface Card

 Hopefully if you are setting up as a reseller you'll learn the diffence
 between an FXS and an FXO, in this case a TDM04B not a TDM40B!
Because you expect resellers to know what they are talking about? That
would be nice, wouldn't it!
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[Asterisk-Users] install e100 card errors

2004-12-13 Thread Jiang zhou

Hi all:

I have install a E100P card. But when load the driver it reports error as
below:

[EMAIL PROTECTED] libpri]# modprobe wct1xxp
/lib/modules/2.4.18-3/misc/wct1xxp.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or IRQ parameters
/lib/modules/2.4.18-3/misc/wct1xxp.o: insmod
/lib/modules/2.4.18-3/misc/wct1xxp.o failed
/lib/modules/2.4.18-3/misc/wct1xxp.o: insmod wct1xxp failed

the pci information as below:

[EMAIL PROTECTED] libpri]# cat /proc/pci
PCI devices found:
  Bus  0, device   0, function  0:
Host bridge: PCI device 8086:2560 (Intel Corp.) (rev 3).
  Prefetchable 32 bit memory at 0xe800 [0xebff].
  Bus  0, device   2, function  0:
VGA compatible controller: PCI device 8086:2562 (Intel Corp.) (rev 3).
  IRQ 12.
  Prefetchable 32 bit memory at 0xe000 [0xe7ff].
  Non-prefetchable 32 bit memory at 0xec10 [0xec17].
  Bus  0, device  29, function  0:
USB Controller: PCI device 8086:24c2 (Intel Corp.) (rev 2).
  IRQ 12.
  I/O at 0xd800 [0xd81f].
  Bus  0, device  29, function  1:
USB Controller: PCI device 8086:24c4 (Intel Corp.) (rev 2).
  IRQ 10.
  I/O at 0xd000 [0xd01f].
  Bus  0, device  29, function  2:
USB Controller: PCI device 8086:24c7 (Intel Corp.) (rev 2).
  IRQ 9.
  I/O at 0xd400 [0xd41f].
  Bus  0, device  29, function  7:
USB Controller: PCI device 8086:24cd (Intel Corp.) (rev 2).
  IRQ 5.
  Non-prefetchable 32 bit memory at 0xec18 [0xec1803ff].
  Bus  0, device  30, function  0:
PCI bridge: Intel Corp. 82820 820 (Camino 2) Chipset PCI (rev 130).
  Master Capable.  No bursts.  Min Gnt=6.
  Bus  0, device  31, function  0:
ISA bridge: PCI device 8086:24c0 (Intel Corp.) (rev 2).
  Bus  0, device  31, function  1:
IDE interface: PCI device 8086:24cb (Intel Corp.) (rev 2).
  IRQ 9.
  I/O at 0x0 [0x7].
  I/O at 0x0 [0x3].
  I/O at 0x0 [0x7].
  I/O at 0x0 [0x3].
  I/O at 0xf000 [0xf00f].
  Non-prefetchable 32 bit memory at 0x1f80 [0x1f8003ff].
  Bus  0, device  31, function  3:
SMBus: PCI device 8086:24c3 (Intel Corp.) (rev 2).
  IRQ 11.
  I/O at 0x5000 [0x501f].
  Bus  1, device   8, function  0:
Ethernet controller: PCI device 8086:103a (Intel Corp.) (rev 130).
  IRQ 11.
  Master Capable.  Latency=32.  Min Gnt=8.Max Lat=56.
  Non-prefetchable 32 bit memory at 0xec00 [0xec000fff].
  I/O at 0xc000 [0xc03f].
  Bus  1, device  13, function  0:
Communication controller: Tiger Jet Network Inc. Model 300 128k (rev 0).
  IRQ 10.
  Master Capable.  No bursts.  Min Gnt=1.Max Lat=128.
  Non-prefetchable 32 bit memory at 0xec001000 [0xec001fff].




Can someone tell how load the driver?

Jiangzhou


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[Asterisk-Users] CVS zaptel missing files

it appears the cvs for zaptel as of 12/13/04 am is missing
at least 1 file -- wcfxs.c
greg
Regards
Greg Cirino
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Re: [Asterisk-Users] CVS zaptel missing files

2004-12-13 Thread Dave Cotton

On Mon, 2004-12-13 at 08:08 -0500, Greg - Cirelle Enterprises wrote:
 it appears the cvs for zaptel as of 12/13/04 am is missing
 at least 1 file -- wcfxs.c

How about wctdm.c ?

-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] What route do diverted SIP calls travel?

2004-12-13 Thread Andy Burns

Sam Bashton wrote:
 The data-heavy portion of the traffic is RTP, and that should be a
 direct connection using your providers gateway.
Thanks, that was what I hoped for, no sense in all the traffic passing 
up and down my ADSL to get back to where it came from, I suppose the 
clue about SIP is in the name, if it only *initiates* the call the 
payload doesn't have to travel the same route as the call setup, nice ;-)

 Make sure you have
 'canreinvite=yes' set in the appropriate section of your sip.conf.
I'll look into that, I'm just getting past the udev issues of asterisk 
on FC3 to get an X100P installed.
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Re: [Asterisk-Users] TDM400P FXS polarity reversal?

2004-12-13 Thread Soren Rathje

Andrew Kohlsmith wrote:
 On December 13, 2004 03:10 am, Soren Rathje wrote:
 wait_just_a_bit(HZ/10);

 I didn't want to wait inside the driver, likely a place where
 interrupts are disabled...

Well, nobody claimed it was ready for production.. :-)  I'm usually OK for
POC code, but don't ask me to do production code, I haven't done any serious
coding the last ~25 years. I usually tell programmers what I want and how I
want it and where to find the bugs...

/Soren

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[Asterisk-Users] Broadvoice Patch Applied to CVS

2004-12-13 Thread Seth Remington

Just in case anybody missed it, the Broadvoice patch has been applied to
CVS HEAD:

=
Sat, 11 Dec 2004 23:33:48 -0600 (CST)

Modified Files:
chan_sip.c 
Log Message:
Merge SIP authentication reuse patch (bug #2917) aka The Broadvoice
Patch with modifications
=

Olle also has an updated patch for CVS stable (1.03) at
http://edvina.net/broadvoice/patch.shtml

-Seth

-- 
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559

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Re: [Asterisk-Users] [OT] Small SIP phones?

2004-12-13 Thread tim panton

Well, I know it isn't a classic style handset, but the Prestige 2000W is
small light and portable. If you have a suitable WiFi Card in your
laptop acting as an access point, you could squeeze a couple into
your laptop bag and do a pretty convincing demo without having to
hook into the client's network at all.
One thing though, the current version of the firmware of these
phones doesn't seem to be able to do 128 bit WEP encryption and
aLaw at the same time, it occasionally drops a 10th of a sec.
I've turned off encryption on that network, and now the phone seems
fine.
Apart, that is, from the fact that I haven't quite got to the bottom of
why it sometimes slips into g729 when I haven't asked it to.
I suspect the fact that I'm running 6 month old * build may be partly 
to blame.

Tim.
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Re: [Asterisk-Users] Strange Segmentation fault

2004-12-13 Thread Paradise Dove

I'm using FC2. but with a fresh 2.6.9 kernel downloaded from kernel.org.
I've recently upgraded my Glibc to glibc-2.3.3-27.1.
I'm also using ECC Reg Memory.
and this is my Xeon CPU info: (HyperThreading is ON)
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 15
model   : 2
model name  : Intel(R) Xeon(TM) CPU 3.20GHz
stepping: 5
cpu MHz : 3199.895
cache size  : 512 KB
physical id : 0
siblings: 2
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr
bogomips: 6340.60



On Mon, 13 Dec 2004 13:58:33 +0100, Andreas Sikkema
[EMAIL PROTECTED] wrote:
 [EMAIL PROTECTED] wrote:
 
 
 
  - dual xeon box (3.2Ghz)
  - 2Gb of memory
  - E7501 chipset motherboard.
  - U320 scsi disks
  - intel Gb ethernet device.
  - i only use sip for clients (no fxs in box)
  - TE405P for fxo (with 4 atran TA750).
  - ulaw is used as codec and echo cancellationo is enabled.
 
  but the core dump file has nothing to show with gdb.
 
  this is the output of gdb:
 
  Program terminated with signal 11, Segmentation fault. #0 
  0xb7fbbce4 in ?? () ___
 
 What Linux (assuming you're running Linux) distribution
 are you running?
 
 I have seen lots of this kind of problems before. We
 had lots of stability problems with GNUgk on Debian
 Woody. Once we moved to Sarge we had no problems at all,
 with uptime going from a couple of days to several
 months when we had no need for GNUgk anymore.
 
 --
 Andreas SikkemaRits tele.com
 Van Vollenhovenstraat 33016 BE Rotterdam
 t: +31 (0)10 2245544f: +31 (0)10 2245540
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Re: [Asterisk-Users] CVS zaptel missing files

2004-12-13 Thread Soren Rathje

Greg - Cirelle Enterprises wrote:
 it appears the cvs for zaptel as of 12/13/04 am is missing
 at least 1 file -- wcfxs.c

It was renamed to wctdm.c around Nov. 6. 2004

/Soren

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Re: [Asterisk-Users] MySQL

2004-12-13 Thread Brian Wilkins

I have found that because the way MyISAM works, InnoDB is a better solution to 
prevent hangups due to the type of locking MyISAM uses. You have to edit 
your /etc/mysql/my.cnf in order to enable InnoDB

From High Performance MySQL by O'Reilly :

innodb_data_file_path = ibdata1:400M
innodb_data_home_dir = /usr/local/mysql/var/
innodb_log_group_home_dir = /usr/local/mysql/var/
innodb_log_arch_dir = /usr/local/mysql/var/
set-variable = innodb_mirrored_log_groups=1
set-variable = innodb_log_files_in_group=3
set-variable = innodb_log_file_size=5M
set-variable = innodb_log_buffer_size=8M
innodb_flush_log_at_trx_commit=1
innodb_log_archive=0
innodb_buffer_pool_size=16M
innodb_additional_mem_pool_size = 2M
 innodb_file_io_threads = 4
innodb_lock_wait_timeout = 50

MyISAM tables, if accessing many different times by many different processes, 
can causes buffering issues; especially on a large scale IP PBX.

Read here on how to convert your MyISAM tables to InnoDB : 
http://dev.mysql.com/doc/mysql/en/Converting_tables_to_InnoDB.html

-- 
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]

Heritage Communications Corporation
  Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net

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RE: [Asterisk-Users] Strange Segmentation fault

2004-12-13 Thread Andreas Sikkema

[EMAIL PROTECTED] wrote:

 - dual xeon box (3.2Ghz)
 - 2Gb of memory
 - E7501 chipset motherboard.
 - U320 scsi disks
 - intel Gb ethernet device.
 - i only use sip for clients (no fxs in box)
 - TE405P for fxo (with 4 atran TA750).
 - ulaw is used as codec and echo cancellationo is enabled.
 
 but the core dump file has nothing to show with gdb.
 
 this is the output of gdb:
 
 Program terminated with signal 11, Segmentation fault. #0 
 0xb7fbbce4 in ?? () ___

What Linux (assuming you're running Linux) distribution 
are you running? 

I have seen lots of this kind of problems before. We 
had lots of stability problems with GNUgk on Debian 
Woody. Once we moved to Sarge we had no problems at all, 
with uptime going from a couple of days to several 
months when we had no need for GNUgk anymore.

-- 
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
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[Asterisk-Users] Asterisk on FreeBSD

2004-12-13 Thread Ali Ziaee





Hi Guys,

I'm very interested if somebody using asterisk on 
FreeBSD and not Linux without problem ?

Thank you for your feedbacks,

Ali
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Re: [Asterisk-Users] MySQL

2004-12-13 Thread Bruce Komito

If you do:

cvs checkout asterisk-addons

(without the -r v1-0), you'll get everything you need...including
res_mysql.conf.sample .

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815

On Mon, 13 Dec 2004, Bill wrote:

Same here. I've deleted and re-installed asterisk a few times and the
RealTime voicemail never works. The best I've gotten is the MySQL query to
execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
libpri asterisk asterisk-addons asterisk-sounds to download the latest
version the res_mysql.conf.sample isn't even there. I made it from scratch
but it still doesn't work. If that file isn't there what else is missing?

Bill

- Original Message -
From: Greg - Cirelle Enterprises
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sunday, December 12, 2004 12:50 PM
Subject: Re: [Asterisk-Users] MySQL

At 06:29 PM 12/9/04, you wrote:
Sure. (I really need to write a wiki on this.)

You have two choices here before we start. You can use RealTime one of 2
ways: ODBC or direct MySQL. Currently these are the only two supported
methods.

Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm
going to instruct on how to use/install it.

The RealTime MySQL driver can be found inside asterisk-addons. Just do the
standard make, make install.

Now copy asterisk-addons/configs/res_mysql.conf.sample to
/etc/asterisk/res_mysql.conf (or whereever your conf dir is).

Edit the res_mysql.conf to your liking.

Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime
config stuff. If you want voicemail, add this line:

voicemail = mysql,asterisk,voicemail_users

No such file res_mysql.conf
only cdr_mysql_conf.sample

Greg

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Re: [Asterisk-Users] voicemail from mysql / change password

Have you examined your debug log for a possible SQL error? Updating user
passwords works fine on our systems. There should be no need to force anyone
to do it a certain way.

-Matthew
- Original Message - 
From: Brad Hughes [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 11, 2004 10:13 AM
Subject: [Asterisk-Users] voicemail from mysql / change password

Im having a problem where I've just switched from static configs to
realtime configs stored in mysql

It's all working fine (in terms of it reading the configs and loading them
as it should), except my problem is that if a user changes there voicemail
password via the Advanced Options (0) in the Voicemail menu via there SIP
phone, the password doesn't get updated in the mysql database (like it used
to in the static voicemail.conf file) - and consequently the next time I
reload asterisk, there voicemail password gets reset back to whatever it
was/is in the mysql database.

Am I overlooking something, or is there an easy solution? If I could just
disable the change password option in Voicemail, that'd be enough for me
(and force them to change it via a web interface). Is that do-able?

Here's the line from my extconfig.conf:

voicemail = mysql,asterisk,users

And the mysql users table schema:

CREATE TABLE users (
   context char(79) DEFAULT '' NOT NULL,
   mailbox char(79) DEFAULT '' NOT NULL,
   password char(79) DEFAULT '' NOT NULL,
   fullname char(79) DEFAULT '' NOT NULL,
   email char(79) DEFAULT '' NOT NULL,
   pager char(79) DEFAULT '' NOT NULL,
   options char(159) DEFAULT '' NOT NULL,
   stamp timestamp,
   PRIMARY KEY (context,mailbox)
);

Thanks

Brad

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RE: [Asterisk-Users] Echo on one E1 line, but not the other

2004-12-13 Thread Asterisk

Both ISDN lines are going into the same * box - span 1 is the test isdn
line and span 3 is the live isdn line. The two ISDN lines are situated
right next to each other!

As mentioned there is no problem with the test line, so there isn't a
problem with * as such (I don't think!). Perhaps I haven't got the
configuration quite right. 

When you say that you adjusted the gain, is that the tx/rx settings in
zapata.conf ? How did you determine the correct settings: by placing a call
and monitoring using ztmonitor ?

Thanks.

Julian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Hanselman
Sent: 13 December 2004 12:13
To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Echo on one E1 line, but not the other

It looks like this is a splice between a couple of ISDN-30 lines and one or
more PBX's?

Are they both with the same provider, or with different providers?

We ended up adjusting the gain our ours as we would hear a distinct echo on
certain calls.

Other than that, you'll need to do the usual tests, check for shared
interrupts, also, see if disk activity causes a problem.

(check /proc/interrupts for shared interrupts).

Have you also checked the output of the pri commands to ensure that you're
not getting line errors?

Steve


-Original Message-
From: Asterisk [mailto:[EMAIL PROTECTED]
Sent: 13 December 2004 12:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Echo on one E1 line, but not the other

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p
EuroISDN.

We have 2 ISDN lines, one we had for testing, and one for general (40+
users) use.

During the testing phase, we had 10 phones linked to the second ISDN line,
and there were no problems with echo at all. Lucky me. However, since we
have started rolling out, we've had quite loud complaints that there is a
terrible echo. If I direct my 7960 to use the primary line, there is an
echo. If I use the second line (dialling the same external number) there is
no echo at all.

What could be the issue ? I have noticed on the primary line there is a
detected rx/tx on channel xx, echo cancellation disabled (or something
like that. Is this the cause ? We have several fax / modems going through
the line - should I always dedicate a channel to them ?

I've included my zaptel.conf and zapata.conf file below. Any help / comments
appreciated.

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg #

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

span=3,2,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78

span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109

loadzone=uk
defaultzone=uk

#

;zapata.conf

[general]

[trunkgroups]

[channels]
musiconhold=default
language=en
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callerid=asreceived
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
;callgroup=1
;pickupgroup=1
;adsi=yes

context=isdn32-b
pridialplan=unknown
group=1
signalling=pri_cpe
switchtype=euroisdn
channel=1-15,17-31

context=meridian-b
group=2
signalling=pri_net
switchtype=euroisdn
channel=32-46,48-62

context=isdn32-a
pridialplan=unknown
group=3
signalling=pri_cpe
switchtype=euroisdn
channel=63-77,79-93

context=meridian-a
group=4
signalling=pri_net
switchtype=euroisdn
channel=94-108,110-124

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Re: [Asterisk-Users] Dialing out to 2 clients simultaneously

2004-12-13 Thread Marc Storck

Hello,
this is not possible,
you will have to solve this via the dialplan using parallel ringing or 
queues.

Regards,
Marc
[EMAIL PROTECTED] wrote:
Hi
When I register a SIP or IAX client to asterisk and I dial to it from
another UA then there is no problem at all 

But, when I register two or more clients to the SAME peer (with the same
user/pass) and I call to this peer.. Then only the UA which registered
the last will ring.. Others don't ring... 

What can I do about this?? 

I would like to register for example 10 UA's to the same peer and want
them all to ring at the same time without having to set up different
usernames and passwords for all these ua's and having to make difficult
dialplans
Is this possible? Am I doing something wrong or is this behaviour by
design?
Regards,
Niels
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Re: [Asterisk-Users] Problems getting Asterisk Realtime to work

It should be [settings] in your extconfig.conf not [default].

-Matthew
- Original Message - 
From: Jason Goecke [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 12, 2004 3:06 AM
Subject: [Asterisk-Users] Problems getting Asterisk Realtime to work


 I have installed the CVS Head as of 12/12/04, as well
 as the asterisk-addons to ensure that
 /usr/lib/asterisk/modules/res_config_mysql.so exists.

 I have configured the following (after building a new
 DB with the appropriate SQL examples, with mods to
 drop the invalid keys, on the Wiki):

 - /etc/asterisk/res_mysql.conf

 [general]
 dbhost = 127.0.0.1
 dbname = my_db
 dbuser = my_uname
 dbpass = my_secret
 dbport = 3306
 dbsock = /tmp/mysql.sock

 - /etc/asterisk/extconfig.conf

 ; Extconfig.conf for realtime configuration

 voicemail = mysql,my_db,voicemail_users

 (Just want to try something simple such as voicemail
 for the initial testing.)

 I have removed the [default] section from my
 voicemail.conf. When I try to access voicemail after
 restarting Asterisk, no voicemail config is found.

 Anyone have any luck?
 -
 I notice I get this error at startup:

 parse error: No category context for line 1 of
 /etc/asterisk/extconfig.conf

 If I change my extconfig.conf to:

 ; Extconfig.conf for realtime configuration

 [default]
 voicemail = mysql,my_db,voicemail_users

 The error goes away, but the config still does not
 work. Can't find anything on the new Wiki pages on the
 subject though.
 -
 Also posted here:


http://asterisk.xvoip.com/viewtopic.php?t=764start=0postdays=0postorder=aschighlight=
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RE: [Asterisk-Users] Echo on one E1 line, but not the other

2004-12-13 Thread Steve Hanselman

More by trial and error, we backed off the gain until it disappeared but
with no detriment to the call quality (didn't want people to sound like a
whisper).

Our situation was somewhat different to yours though, we were seeing the
issues on calls from our PBX, not on calls through the IP phones.

Who hears the echo, the IP phone user or the remote user?

Steve


-Original Message-
From: Asterisk [mailto:[EMAIL PROTECTED] 
Sent: 13 December 2004 12:36
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Echo on one E1 line, but not the other

Both ISDN lines are going into the same * box - span 1 is the test isdn
line and span 3 is the live isdn line. The two ISDN lines are situated
right next to each other!

As mentioned there is no problem with the test line, so there isn't a
problem with * as such (I don't think!). Perhaps I haven't got the
configuration quite right. 

When you say that you adjusted the gain, is that the tx/rx settings in
zapata.conf ? How did you determine the correct settings: by placing a call
and monitoring using ztmonitor ?

Thanks.

Julian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Hanselman
Sent: 13 December 2004 12:13
To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Echo on one E1 line, but not the other

It looks like this is a splice between a couple of ISDN-30 lines and one or
more PBX's?

Are they both with the same provider, or with different providers?

We ended up adjusting the gain our ours as we would hear a distinct echo on
certain calls.

Other than that, you'll need to do the usual tests, check for shared
interrupts, also, see if disk activity causes a problem.

(check /proc/interrupts for shared interrupts).

Have you also checked the output of the pri commands to ensure that you're
not getting line errors?

Steve


-Original Message-
From: Asterisk [mailto:[EMAIL PROTECTED]
Sent: 13 December 2004 12:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Echo on one E1 line, but not the other

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p
EuroISDN.

We have 2 ISDN lines, one we had for testing, and one for general (40+
users) use.

During the testing phase, we had 10 phones linked to the second ISDN line,
and there were no problems with echo at all. Lucky me. However, since we
have started rolling out, we've had quite loud complaints that there is a
terrible echo. If I direct my 7960 to use the primary line, there is an
echo. If I use the second line (dialling the same external number) there is
no echo at all.

What could be the issue ? I have noticed on the primary line there is a
detected rx/tx on channel xx, echo cancellation disabled (or something
like that. Is this the cause ? We have several fax / modems going through
the line - should I always dedicate a channel to them ?

I've included my zaptel.conf and zapata.conf file below. Any help / comments
appreciated.

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg #

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

span=3,2,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78

span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109

loadzone=uk
defaultzone=uk

#

;zapata.conf

[general]

[trunkgroups]

[channels]
musiconhold=default
language=en
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callerid=asreceived
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
;callgroup=1
;pickupgroup=1
;adsi=yes

context=isdn32-b
pridialplan=unknown
group=1
signalling=pri_cpe
switchtype=euroisdn
channel=1-15,17-31

context=meridian-b
group=2
signalling=pri_net
switchtype=euroisdn
channel=32-46,48-62

context=isdn32-a
pridialplan=unknown
group=3
signalling=pri_cpe
switchtype=euroisdn
channel=63-77,79-93

context=meridian-a
group=4
signalling=pri_net
switchtype=euroisdn
channel=94-108,110-124

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Re: [Asterisk-Users] MySQL

Get newest CVS. Its in there. Trust me. Oh..be sure your getting
asterisk-addons.

-Matthew
- Original Message - 
From: Greg - Cirelle Enterprises [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, December 12, 2004 12:50 PM
Subject: Re: [Asterisk-Users] MySQL

 At 06:29 PM 12/9/04, you wrote:
 Sure. (I really need to write a wiki on this.)

 You have two choices here before we start. You can use RealTime one of 2
 ways: ODBC or direct MySQL. Currently these are the only two supported
 methods.

 Since I don't use ODBC and as the author of the MySQL RealTime driver,
I'm
 going to instruct on how to use/install it.

 The RealTime MySQL driver can be found inside asterisk-addons. Just do
the
 standard make, make install.

 Now copy asterisk-addons/configs/res_mysql.conf.sample to
 /etc/asterisk/res_mysql.conf (or whereever your conf dir is).

 Edit the res_mysql.conf to your liking.

 Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime
 config stuff. If you want voicemail, add this line:

 voicemail = mysql,asterisk,voicemail_users

 No such file res_mysql.conf
 only cdr_mysql_conf.sample

 Greg

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Re: [Asterisk-Users] MySQL

You are missing the fact that RealTime is not 1-0, its CVS. 'Thats' why
res_mysql.conf isn't even there.

-Matthew

- Original Message - 
From: Bill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, December 13, 2004 8:32 AM
Subject: Re: [Asterisk-Users] MySQL

 Same here. I've deleted and re-installed asterisk a few times and the
 RealTime voicemail never works. The best I've gotten is the MySQL query to
 execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
 libpri asterisk asterisk-addons asterisk-sounds to download the latest
 version the res_mysql.conf.sample isn't even there. I made it from
scratch
 but it still doesn't work. If that file isn't there what else is missing?

   Bill

 - Original Message - 
 From: Greg - Cirelle Enterprises
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Sent: Sunday, December 12, 2004 12:50 PM
 Subject: Re: [Asterisk-Users] MySQL

 At 06:29 PM 12/9/04, you wrote:
 Sure. (I really need to write a wiki on this.)

 You have two choices here before we start. You can use RealTime one of 2
 ways: ODBC or direct MySQL. Currently these are the only two supported
 methods.

 Since I don't use ODBC and as the author of the MySQL RealTime driver,
I'm
 going to instruct on how to use/install it.

 The RealTime MySQL driver can be found inside asterisk-addons. Just do
the
 standard make, make install.

 Now copy asterisk-addons/configs/res_mysql.conf.sample to
 /etc/asterisk/res_mysql.conf (or whereever your conf dir is).

 Edit the res_mysql.conf to your liking.

 Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime
 config stuff. If you want voicemail, add this line:

 voicemail = mysql,asterisk,voicemail_users

 No such file res_mysql.conf
 only cdr_mysql_conf.sample

 Greg

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Re: [Asterisk-Users] CVS zaptel missing files

At 08:19 AM 12/13/04, you wrote:
On Mon, 2004-12-13 at 08:08 -0500, Greg - Cirelle Enterprises wrote:
 it appears the cvs for zaptel as of 12/13/04 am is missing
 at least 1 file -- wcfxs.c
How about wctdm.c ?
--
Dave Cotton [EMAIL PROTECTED]

Not sure what that is supposed to do but it
sure don't do the trick out of the box.
To get the tdm and t100 cards to light up
i have to revert to ver 101, 102, or 103 of
zaptel
greg
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Re: [Asterisk-Users] TDM400P FXS polarity reversal?

2004-12-13 Thread Andrew Kohlsmith

On December 13, 2004 08:29 am, Soren Rathje wrote:
 Well, nobody claimed it was ready for production.. :-)  I'm usually OK for
 POC code, but don't ask me to do production code, I haven't done any
 serious coding the last ~25 years. I usually tell programmers what I want
 and how I want it and where to find the bugs...

That is pretty much exactly how I work too...  Damn, too many chiefs and not 
enough indians.  :-)

-A.
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Re: [Asterisk-Users] MySQL

2004-12-13 Thread VCI Help Desk

What's the proper way to download a STABLE version of asterisk and
asterisk-addons from CVS? I keep finding documentation that says two
different ways of download it.

Now that I've downloaded the asterisk-addons that has the
res_mysql.conf.sample it won't compile. If I cd to asterisk-addons and do
a make clean; make I get the following. This used to work fine before.

res_config_mysql.c: In function `load_module':
res_config_mysql.c:467: error: structure has no member named `static_func'
res_config_mysql.c:468: error: structure has no member named `realtime_func'
res_config_mysql.c:469: error: structure has no member named `update_func'
res_config_mysql.c:470: error: structure has no member named
`realtime_multi_func'
make: *** [res_config_mysql.o] Error 1
rm app_saycountpl.o

The mysql-vm-routines.h is still there as well. I thought that file
was removed.

  Bill




- Original Message - 
From: Matthew Boehm
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, December 13, 2004 9:09 AM
Subject: Re: [Asterisk-Users] MySQL


You are missing the fact that RealTime is not 1-0, its CVS. 'Thats' why
res_mysql.conf isn't even there.

-Matthew

- Original Message - 
From: Bill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, December 13, 2004 8:32 AM
Subject: Re: [Asterisk-Users] MySQL


 Same here. I've deleted and re-installed asterisk a few times and the
 RealTime voicemail never works. The best I've gotten is the MySQL query to
 execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
 libpri asterisk asterisk-addons asterisk-sounds to download the latest
 version the res_mysql.conf.sample isn't even there. I made it from
scratch
 but it still doesn't work. If that file isn't there what else is missing?

   Bill





 - Original Message - 
 From: Greg - Cirelle Enterprises
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Sent: Sunday, December 12, 2004 12:50 PM
 Subject: Re: [Asterisk-Users] MySQL


 At 06:29 PM 12/9/04, you wrote:
 Sure. (I really need to write a wiki on this.)
 
 You have two choices here before we start. You can use RealTime one of 2
 ways: ODBC or direct MySQL. Currently these are the only two supported
 methods.
 
 Since I don't use ODBC and as the author of the MySQL RealTime driver,
I'm
 going to instruct on how to use/install it.
 
 The RealTime MySQL driver can be found inside asterisk-addons. Just do
the
 standard make, make install.
 
 Now copy asterisk-addons/configs/res_mysql.conf.sample to
 /etc/asterisk/res_mysql.conf (or whereever your conf dir is).
 
 Edit the res_mysql.conf to your liking.
 
 Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime
 config stuff. If you want voicemail, add this line:
 
 voicemail = mysql,asterisk,voicemail_users

 No such file res_mysql.conf
 only cdr_mysql_conf.sample

 Greg

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[Asterisk-Users] only allow long distance calls to countries x, y, and z

2004-12-13 Thread Thomas Miller

Can somebody suggest theeasiestway to only allow outgoing long distance calls to countries x, y, and z? 

Since Broadvoice allows free long distance to a bunch of countries I would like to take advantage of that, but block all other long distance calls.

Thanks,
Tom
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[Asterisk-Users] Re: Dialing out to 2 clients simultaneously

2004-12-13 Thread Tom Ivar Helbekkmo

[EMAIL PROTECTED] writes:

 This is the last issue I have which makes that I can't get rid of the
 SER proxy in front of asterisk.. Want to get rid of it

Out of curiosity, why?

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
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RE: [Asterisk-Users] Re: Dialing out to 2 clients simultaneously

2004-12-13 Thread niels


Because else people will complain that they can't register two
softphones anymore with same user/pass (because only one of the two
softphones can receive the incoming calls) :-)



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Ivar
Helbekkmo
Sent: Monday, December 13, 2004 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Dialing out to 2 clients simultaneously

[EMAIL PROTECTED] writes:

 This is the last issue I have which makes that I can't get rid of the 
 SER proxy in front of asterisk.. Want to get rid of it

Out of curiosity, why?

-tih
--
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
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Re: [Asterisk-Users] RE: Polycom 500 - Dialtone while connected

2004-12-13 Thread Jorge Mendoza

Greg Boehnlein wrote:
On Thu, 9 Dec 2004, Jorge Mendoza wrote:

Andrei,
I'm interested too. Any chance to put the archive in a ftp site?.
Jorge Mendoza

I am also interested in getting the 1.3.4 firmware. It annoys me that I 
can't just get it from Polycom's website, and forces me to rethink 
deploying their phones for customers.

Some time ago, somebody in the list contacted a Polycom manager and the 
result was (if I'm not wrong) the site:
http://www.freedomphones.net/polycom/files/
It is supposed to be updated, but it missing the last firmware.
A people kindly sent me the 1.3.4 firmware and I would like to upload it 
to the site mentioned before. I do not if the webmaster or the 
maintainer would like to do the job, or there are legal problems?

Jorge Mendoza
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[Asterisk-Users] How to create a confrence using SIP channels

2004-12-13 Thread Bartosz Wegrzyn - asterisk

Hello,

I would like to be able to dial in to my asterisk box.
Dial extension which would call two other people using the Sip channels.
We would like to be able to talk to each other at the same time.

Thanks

Bartosz Wegrzyn

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Re: [Asterisk-Users] MySQL

Are there any others besides CVS and STABLE.

No. Only those 2. Unless I'm mistaken.

 When someone downloads using cvs checkout -r v1-0  what version
is
 that, CVS or stable?

The 1.0.* branch is refered to as STABLE. Anything above that, is called
CVS. (Also called CVS-HEAD)

-Matthew

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Re: [Asterisk-Users] four wildcards in a single pc

2004-12-13 Thread Gilad Ben-Yossef

Hi Jim,
Jim Van Meggelen wrote:

Getting dedicated IRQs for the cards is a minor problem compared to what
happens when you have four cards hammering away mercilessly at the
chipset and CPU of your motherboard; 1000 IRQs per second, per card.
Nobody's really sure what's wrong, but it causes problems for pretty
nearly everyone.
From your description it is very clear what is wrong - the machine is 
heavily over loaded (or sometime having load spikes) due to interrupt 
livelock. It spends so much resources dealing with interrupts that it 
doesn't have enough CPU time to handle any thing else.

If anyone is interested in a very more info about this phenomena, simply 
search google for interrupt livelock and interrupt mitigation. Most 
of the research pertaining to this problem was done for network cards 
but it really applies to any source of (too many) interrupts.

I've had some expereicne dealing with the problem in network cards. If I 
can help in any way...

Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com  | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax:  +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
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Re: [Asterisk-Users] How to create a confrence using SIP channels

2004-12-13 Thread Peter Svensson

On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:

 I would like to be able to dial in to my asterisk box.
 Dial extension which would call two other people using the Sip channels.
 We would like to be able to talk to each other at the same time.

This is quite easy. :-)

Have the extension construct two call files and place them in the outgoing
call spool directory then proceed to the meetme room. The call files
should contain the meetme extension as well. Thus the caller will go to
the meetme conference directly and the two called parties will enter the 
conference as soon as they answer.

Peter


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Re: [Asterisk-Users] Asterisk on SuSE 9.1?

2004-12-13 Thread Vassilis Konstantinou

I compiled version 1.0.3 over teh weekend on a Suse 9.1 box. It was a clean 
installation straight out of the SUSE cds. Make sure that the kernel 
sources are loaded and that you do a full online update before you proceed. 
Asterisk compiles without any problems.

Vassilis
At 17:44 13/12/2004, you wrote:
I am trying to do my first asterisk install on a SuSE 9.1 box, using the
asterisk-update script mentioned a few days ago on this list.
  I did read the 'quickstart' document on onlamp.com, and made sure the
following packages were installed via yast:
 bison, cvs, gcc, kernel-source, libtermcap-devel, ncurses-devel,
newt-devel, openssl096b, and openssl-devel.
The SuSE 9.1 DVD contained openssl-0.9.7d-15.  I hope that's compatible,
since its 'later' than 096b, right?
...there was a termcap package, which contains the termcap libraries, but
no libtermcap-devel.  If there are header files necessary in addition to
the libraries, does someone know where I can obtain them packaged for SuSE
9.1?
After pulling down all the source from CVS, the script begins compiling.
It finds a 2.6 kernel, and tells me that kernel sources aren't necessary
with 2.6 kernels.  It then proceeds to compile several modules, but quits
when the compile of zttest returns a message telling me that I need kernel
sources in order to compile.
Can someone point me in the right direction?
--
Rick Green
They that can give up essential liberty to obtain a little
 temporary safety, deserve neither liberty nor safety.
  -Benjamin Franklin
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[Asterisk-Users] Re: Asterisk on SuSE 9.1?

2004-12-13 Thread Don Hughes



On 13 Dec 2004 at 12:44, Rick Green wrote:

 I am trying to do my first asterisk install on a SuSE 9.1 box, using
 the asterisk-update script mentioned a few days ago on this list.
 
   I did read the 'quickstart' document on onlamp.com, and made sure
   the
 following packages were installed via yast:
  bison, cvs, gcc, kernel-source, libtermcap-devel, ncurses-devel,
 newt-devel, openssl096b, and openssl-devel.
 

I am using SuSE 9.1 with:

bison-1.875-51.4
cvs-1.11.14.26.6
gcc-3.3.3-41
kernel-source-2.6.5-7.11.5
termcap-2.0.8-876
ncurses-5.4-61.3
ncurses-devel-5.4-61.3
openssl-0.9.7d-15.13
openssl-devel-0.9.7d-15


..don

support at microtechniques.com
White Plains, NY


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Re: [Asterisk-Users] Re: four wildcards in a single pc

2004-12-13 Thread Andrew Kohlsmith

On December 13, 2004 11:53 am, Stephen R. Besch wrote:
 1) This is not to minimize the problem, but 1000 interrupts per second
 is quite a few, but not an overwhelming amount. Keep in mind that an
 unbuffered serial card (and there are more than a few of these out
 there) working at 19.2 Kbaud will rack up 1900 interrupts per second and
 the CPU doesn't even sweat. Even an old pig CPU wasn't much strained by

Ahh but the dumb serial driver isn't trying to do echo cancellation or other 
CPU intensive tasks in the bottom-half interrupt handler (i.e. when 
interrupts are disabled) -- the Digium drivers I think are unique in this 
respect.

 2) While it is hard to estimate directly, if the driver is properly
 designed, the number of interrupts should not scale linearly. One drive
 should handle all cards, and part of the time there will be more than
 one card needing servicing on an interrupt. If the driver does not test
 for this, then it should.

I've brought this up a few times -- If you have 1 1000Hz interrupt already, 
use it and disable the timer on subsequent cards and use the one interrupt to 
start capturing and working with the data on ALL cards.

   In fact, I suspect that the buffer is a standard 16-byte FIFO with the
 threshold set at 8 bytes. What this means is that it would be possible
 to handle all 4 cards in a single interrupt, depending of course on the
 design of the buffer, by always emptying all 4 card's buffers on any
 interrupt. In fact, only one card (i.e., the master) should even have
 interrupts enabled!

IIRC The cards do not have 16550-style UARTs on them, so the FIFOs could be 
anything.

 4) All of the preceding notwithstanding, I suspect that the real issue
 has nothing (or little) to do with interrupt load, but, given that the
 card uses CPU cycles rather than a DSP, the problem is more likley CPU
 overload from data handling, which in turn, causes missed interrupts.

I disagree -- I've been doing some (very basic) preliminary testing with the 
Sangoma A101u card (single/dual-span T1) and even with echo cancellation the 
card seems to be far more able to handle shared interrupts and high CPU loads 
without sounding funny -- perhaps this is just some kind of driver issue, 
since really hardware's hardware and 1000 interrupts a second is an eon to a 
modern CPU.

-A.
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[Asterisk-Users] Asterisk and Sipura SPA-2000

2004-12-13 Thread dkidwell

Hello all,

So i am new to asterisk and very green when it comes to Linux, so don't beat on 
me too bad :)

I just set up * on Red Hat 9.0 last night... everything seems to be 
configured coffectly, I can start * no problem and get the CLI prompt... now 
here is my question... I have an account set up with VoicePulse Connect! and 
also have a Sipura Spa-2000... i am trying to get make it so that the spa-2000 
talks to * which talks to VoicePulse Connect...

So here is where I am at... I have the spa-2000 talking to asterisk... the Line 
1 shows it registered to asterisk... I have daltone... and thats about it... 

I tried to follow the example listed below:

http://voxilla.com/modules.php?op=modloadname=Newsfile=articlesid=39

evertthing went smooth till I tried to dial the examples #'s

when I dialed 786, I did see CID, CIDNAME, DIAL excute... I get error I'm 
sorry that is not a valid ext

when trying to dial 612125551212, I just get a fast busy

I also cant dial any other numbers... 

I have double checked my settings and all looks well, so I thought i'd give the 
list a shot :)

Thanks in Advance!!
David

btw i have learned a TON just by reading your questions and answers!! awesome 
group!!


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[Asterisk-Users] Can a TDM21 and a X100P co-exist

2004-12-13 Thread Vassilis Konstantinou

Well.. subject says it all really.
I have  a TDM with 2 FXS modules and 1 FXO and a X100P.
If I load teh zaptel and wctdm drivers. Asterisk sees the TDM ports fine 
but not the X100P
I have tried several combinations of port numbering but can some kind 
person with a similar setup to send me the correct zaptel.conf, zapata.conf 
and which drivers to load with modprobe? many thanks

Vassilis 

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Re: [Asterisk-Users] Asterisk on FreeBSD

 I'm very interested if somebody using asterisk on 
 FreeBSD and not Linux without problem ? 

I think a lot depends on whether you need hardware interfaces or not.
For voIP only I had no problem on FreebSD 4.5
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[Asterisk-Users] DTMF

2004-12-13 Thread Robin van Leyden

Hello all,

Is it possible to send dtmf tones to an answering terminal (after answering
the call)? 

I have  for example a external voicemail system that I want to connect to *.
Now for the right integration I need to send dtmf tones to the analog ports
that answered the call.

Cheers.

Robin
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[Asterisk-Users] Multiline / Console / Receptionist phone

2004-12-13 Thread Gerald J. Puhl





I have been looking to see if this type of phone can be implimented in
*. I have found nothing conclusive. Is any out there using a
multiline / mutlifunction phone typically used by a receptionist for
transfering / routing calls? I need to know how this is accomplished
or what alternative exists for this.

Thanx!
Gary P.
-- 
Signature




  

  Prototypes
  
  Patterns
  
  Models
  
  Dies
  
  Fixtures
  

  


Please visit www.jppattern.com for more
information about J.P. Pattern, Inc.




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Re: [Asterisk-Users] CallerID after Supervised Transfer

2004-12-13 Thread Eldon Balzer

I am having a similar problem for my home setup.  I receive a call from
the PSTN and have * automatically dial one or both Cisco 7940's running
SIP firmware.  In my case the callerid info just says asterisk as the
PBX is actually placing the call on behalf of the incoming PSTN call on an
X100P adapter.


-= EB =-

 Is there a way to keep the incoming CallerID from the PSTN and pass it
 onto the sip phone receiving the supervised call transfer?



 The receptionist receives the PSTN callerID, performs a supervised
 transfer, we get her local SIP callerID, not the original callers.



 The main reason we would like the true callerID is for asterisk monitor
 to name the file correctly for call records.



 Is this possible with Asterisk?
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Re: [Asterisk-Users] Asterisk and Cisco 7905G or Cisco 7912G

2004-12-13 Thread Adi Linden

 I can't speak for the 7912G, but I have several 7905G phones and these
 work perfectly with Asterisk.

This is great! The 7905G is what I have in mind for a plain basic phone
and the 7940 where a speaker phone is needed.

 The firmware is easy to obtain if you have a Cisco support agreement -
 it's downloadable from CCO (the 7905G and 7912G have different
 firmware builds, but a similar configuration process - be aware of
 that).

Cisco support for the phones is certainly on the reasonable side,
especially since it includes hardware replacement, not just software
support.

Now all I need to find is a low-cost switch that supports PoE for Cisco
phones. A 3550 for my home is very unreasonable, IMHO.

Adi
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[Asterisk-Users] Voicemail and MySQL

2004-12-13 Thread Bill

I have Asterisk talking to MySQL using Realtime but for some reason I
keep getting the wrong context used when Realtime makes the MySQL call. I
can see this in my /var/log/mysql.log file. Because of this I can't login to
VoicemailMain from my X-Ten phone. I can login if I statically configure the
voicemail user in voicemail.conf but I prefer the MySQL.

SELECT * FROM users WHERE mailbox = '0063' AND context = 'default'

In my sip.conf file I have the default settings except the default
context is set. I removed all the example SIP configs further down the
config.

[general]
context=from-sip ;Default context for incoming calls

In my extensions.conf file I have the following. All example extension
configs have been removed.

[from-sip]
exten = 8500,1,VoicemailMain
exten = 8500,n,Hangup

What am I doing wrong?


  Bill

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Re: [Asterisk-Users] Re: four wildcards in a single pc

2004-12-13 Thread Gregory Junker

It would be nice for Mark to comment on this design flaw ...?
Why so quick to assume it's a flaw? Perhaps it's a compromise.
Greg
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[Asterisk-Users] Incoming Toll-Free

2004-12-13 Thread Mark Halverson

Sorry if this is the wrong list...

I need a toll-free number to be delivered to me on IAX. (This is NOT an
existing number need to buy the whole service.)

Anyone know of a service provider offering this?

-Mark
707-735-1038

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Re: [Asterisk-Users] Incoming Toll-Free

2004-12-13 Thread Christopher L. Wade

Mark Halverson wrote:
Sorry if this is the wrong list...
I need a toll-free number to be delivered to me on IAX. (This is NOT an
existing number need to buy the whole service.)
Anyone know of a service provider offering this?
-Mark
707-735-1038
ask this on the -biz list.
-Chris
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administratordba Sparco.com
Email: [EMAIL PROTECTED] 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053
Fax:   (901) 872 8482  USA
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Re: [Asterisk-Users] Voicemail and MySQL

A..now we find the problem.
Voicemailmain does NOT use the context that calls it.
haha..finally found the problem.

You must call it as VoicemailMain(@from-sip) if you want it to look for
mailbox in a specific context.

I knew it was something simple.

-Matthew

- Original Message - 
From: Bill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, December 13, 2004 1:04 PM
Subject: [Asterisk-Users] Voicemail and MySQL


 I have Asterisk talking to MySQL using Realtime but for some reason I
 keep getting the wrong context used when Realtime makes the MySQL call. I
 can see this in my /var/log/mysql.log file. Because of this I can't login
to
 VoicemailMain from my X-Ten phone. I can login if I statically configure
the
 voicemail user in voicemail.conf but I prefer the MySQL.

 SELECT * FROM users WHERE mailbox = '0063' AND context = 'default'

 In my sip.conf file I have the default settings except the default
 context is set. I removed all the example SIP configs further down the
 config.

 [general]
 context=from-sip ;Default context for incoming calls

 In my extensions.conf file I have the following. All example extension
 configs have been removed.

 [from-sip]
 exten = 8500,1,VoicemailMain
 exten = 8500,n,Hangup

 What am I doing wrong?


   Bill

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[Asterisk-Users] weird ring behavior

2004-12-13 Thread Ryan Stark

In my queue I have about 4 agents answering at any given time, * has a 
tendency of rininging the first agent (rrmemory) for only half a ring 
then moving to the next agent, on the console it says it tried them for 
20seconds.  Anyone seen this or know where to look to fix it?

Thanks,
-Ryan
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Re: [Asterisk-Users] Can a TDM21 and a X100P co-exist

2004-12-13 Thread Steven Critchfield

On Mon, 2004-12-13 at 18:47 +, Vassilis Konstantinou wrote:
 Well.. subject says it all really.
 I have  a TDM with 2 FXS modules and 1 FXO and a X100P.
 If I load teh zaptel and wctdm drivers. Asterisk sees the TDM ports fine 
 but not the X100P
 I have tried several combinations of port numbering but can some kind 
 person with a similar setup to send me the correct zaptel.conf, zapata.conf 
 and which drivers to load with modprobe? many thanks

wctdm is for the tdm400 card only. After you modprobe it, modprobe the
wcfxo driver and then your x100P will show up. Remember the order you
load the drivers will determine the oder they are numbered.

-- 
Steven Critchfield [EMAIL PROTECTED]

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[Asterisk-Users] Re: Asterisk on FreeBSD

2004-12-13 Thread Randy Bush

 I'm very interested if somebody using asterisk on FreeBSD and not Linux
 without problem ?

many of us are using * on 5.3-stable and 6.0-current.  without a
problem would be a bit pollyanna-like.

randy

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Re: [Asterisk-Users] MYSQL cmd - preconnect?

Check out RealTime. This is how its done.

Otherwise..not really..each module has its own memory space and runs in its
own thread and so you can't share resources like that across memory space.

What are you using to query?

-Matthew

- Original Message - 
From: Roy Sigurd Karlsbakk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, December 13, 2004 6:42 AM
Subject: [Asterisk-Users] MYSQL cmd - preconnect?


 hi

 is it possible to have asterisk connect to mysql with a
 username/password in some config file and then, afterwards, just use a
 global handle to the db? I don't see the point of connecting every time
 I need to query it ...

 roy

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Re: [Asterisk-Users] MySQL

At 09:32 AM 12/13/04, you wrote:
Same here. I've deleted and re-installed asterisk a few times and the
RealTime voicemail never works. The best I've gotten is the MySQL query to
execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
libpri asterisk asterisk-addons asterisk-sounds to download the latest
version the res_mysql.conf.sample isn't even there. I made it from scratch
but it still doesn't work. If that file isn't there what else is missing?
  Bill

I just found out (on my system), the res_mysql.conf has the
local mysql socket setting looking for mysql.sock in /tmp/mysql.sock
I did a locate mysql.sock which found the actual location and I
put that location in res_mysql.conf in the dbsock parameter and
it began working.
Hope this helps you
Greg
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Re: [Asterisk-Users] MySQL

2004-12-13 Thread Brian Wilkins

Yep, same problem I had. Look in /etc/mysql/my.cnf for the location of the 
sock file.


On Monday 13 December 2004 03:17 pm, Greg - Cirelle Enterprises wrote:
 At 09:32 AM 12/13/04, you wrote:
  Same here. I've deleted and re-installed asterisk a few times and the
 RealTime voicemail never works. The best I've gotten is the MySQL query to
 execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
 libpri asterisk asterisk-addons asterisk-sounds to download the latest
 version the res_mysql.conf.sample isn't even there. I made it from
  scratch but it still doesn't work. If that file isn't there what else is
  missing?
 
Bill

 I just found out (on my system), the res_mysql.conf has the
 local mysql socket setting looking for mysql.sock in /tmp/mysql.sock

 I did a locate mysql.sock which found the actual location and I
 put that location in res_mysql.conf in the dbsock parameter and
 it began working.

 Hope this helps you

 Greg

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-- 
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]

Heritage Communications Corporation
  Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net

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RE: [Asterisk-Users] four wildcards in a single pc

2004-12-13 Thread Grady Trew, Jr.

 Getting dedicated IRQs for the cards is a minor problem compared to what
 happens when you have four cards hammering away mercilessly at the
 chipset and CPU of your motherboard; 1000 IRQs per second, per card.
 Nobody's really sure what's wrong, but it causes problems for pretty
 nearly everyone.

Would the same issues arise with the use of a single Voicetronix 12 port
card?  What about using 2 of them in the same machine?


Thanks

Grady


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Re: [Asterisk-Users] MySQL

Even though you can...why would you? You can't use some things that are in
CVS addons with STABLE asterisk.

res_config_mysql.c and res_mysql.conf are part of the CVS version of
asterisk. This means that you cannot use them with STABLE.

If you want RealTime functionality you HAVE to upgrade your entire asterisk
code to CVS.

-Matthew

- Original Message - 
From: VCI Help Desk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, December 13, 2004 9:21 AM
Subject: Re: [Asterisk-Users] MySQL


 What's the proper way to download a STABLE version of asterisk and
 asterisk-addons from CVS? I keep finding documentation that says two
 different ways of download it.

 Now that I've downloaded the asterisk-addons that has the
 res_mysql.conf.sample it won't compile. If I cd to asterisk-addons and
do
 a make clean; make I get the following. This used to work fine before.

 res_config_mysql.c: In function `load_module':
 res_config_mysql.c:467: error: structure has no member named `static_func'
 res_config_mysql.c:468: error: structure has no member named
`realtime_func'
 res_config_mysql.c:469: error: structure has no member named `update_func'
 res_config_mysql.c:470: error: structure has no member named
 `realtime_multi_func'
 make: *** [res_config_mysql.o] Error 1
 rm app_saycountpl.o

 The mysql-vm-routines.h is still there as well. I thought that file
 was removed.

   Bill




 - Original Message - 
 From: Matthew Boehm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Sent: Monday, December 13, 2004 9:09 AM
 Subject: Re: [Asterisk-Users] MySQL


 You are missing the fact that RealTime is not 1-0, its CVS. 'Thats' why
 res_mysql.conf isn't even there.

 -Matthew

 - Original Message - 
 From: Bill [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
 Sent: Monday, December 13, 2004 8:32 AM
 Subject: Re: [Asterisk-Users] MySQL


  Same here. I've deleted and re-installed asterisk a few times and
the
  RealTime voicemail never works. The best I've gotten is the MySQL query
to
  execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
  libpri asterisk asterisk-addons asterisk-sounds to download the latest
  version the res_mysql.conf.sample isn't even there. I made it from
 scratch
  but it still doesn't work. If that file isn't there what else is
missing?
 
Bill
 
 
 
 
 
  - Original Message - 
  From: Greg - Cirelle Enterprises
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Sent: Sunday, December 12, 2004 12:50 PM
  Subject: Re: [Asterisk-Users] MySQL
 
 
  At 06:29 PM 12/9/04, you wrote:
  Sure. (I really need to write a wiki on this.)
  
  You have two choices here before we start. You can use RealTime one of
2
  ways: ODBC or direct MySQL. Currently these are the only two supported
  methods.
  
  Since I don't use ODBC and as the author of the MySQL RealTime driver,
 I'm
  going to instruct on how to use/install it.
  
  The RealTime MySQL driver can be found inside asterisk-addons. Just do
 the
  standard make, make install.
  
  Now copy asterisk-addons/configs/res_mysql.conf.sample to
  /etc/asterisk/res_mysql.conf (or whereever your conf dir is).
  
  Edit the res_mysql.conf to your liking.
  
  Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the
RealTime
  config stuff. If you want voicemail, add this line:
  
  voicemail = mysql,asterisk,voicemail_users
 
  No such file res_mysql.conf
  only cdr_mysql_conf.sample
 
  Greg
 
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[Asterisk-Users] Detect line in use?

2004-12-13 Thread Jared Armstrong









Is there anyway to determine if is line is already in use by
another device such as a fax machine if the fax machine is not tunneled through
asterisk via a FXS out on a FXO? Right now * tries to pickup the line and dial
when I try the Channel Available command on the Zap FXO. Also, is there a way
to tell * to not do anything or close the connection if it detects a Fax,
without sending the Hangup command, so that the fax machine on the line can
receive properly?



Jared Armstrong






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RE: [Asterisk-Users] Pitching Asterisk

2004-12-13 Thread Damon Estep

http://www.millenigence.com/articles/asterisk-non-technical-review.pdf 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Sean Cook
 Sent: Monday, December 13, 2004 8:10 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Pitching Asterisk

 The company I work for is looking at vendors for a PBX, one 
 of the requirements is VoIP.  I have been sitting there 
 listening to people pitch very proprietary implementations of 
 VoIP where you are locked in to their hardware, their interface...

 I know a little bit about asterisk (set up a couple offices with it...
 run it at home...) and would like to pitch it to this 
 company.  Does someone have a decent presentation that I 
 could use as a starting point?
 Basically I am looking for a business oriented (not too 
 technical) overview of asterisk, or asterisk for suits.  

 any help would be appreciated.

 Sean

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Re: [Asterisk-Users] Pitching Asterisk

2004-12-13 Thread Jason Becker

Sean Cook wrote:
The company I work for is looking at vendors for a PBX, one of the
requirements is VoIP.  I have been sitting there listening to people
pitch very proprietary implementations of VoIP where you are locked in
to their hardware, their interface...
I know a little bit about asterisk (set up a couple offices with it...
run it at home...) and would like to pitch it to this company.  Does
someone have a decent presentation that I could use as a starting point?
Basically I am looking for a business oriented (not too technical)
overview of asterisk, or asterisk for suits.  
http://graphics.cs.uni-sb.de/VCORE/Publications/mark_spencer/mark.smil
A presentation by Mark Spencer that discusses the Business and Technical 
Details of Asterisk.

Regards,
--
Jason Becker
Director  CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
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Re: [Asterisk-Users] four wildcards in a single pc

2004-12-13 Thread mike+asterisk-users

On Mon, Dec 13, 2004 at 03:15:18PM -0600, Grady Trew, Jr. wrote:
  Getting dedicated IRQs for the cards is a minor problem compared to what
  happens when you have four cards hammering away mercilessly at the
  chipset and CPU of your motherboard; 1000 IRQs per second, per card.
  Nobody's really sure what's wrong, but it causes problems for pretty
  nearly everyone.
 
 Would the same issues arise with the use of a single Voicetronix 12 port
 card?  What about using 2 of them in the same machine?

This would be rather silly as you'd be better off price wise getting a
single T1 card and a channel bank. 

Otherwise, the Voicetronix board may or may not even do the 1k/s.


-- 
Mike Mattice - Systems Programmer and Administrator
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[Asterisk-Users] phpconfig - can't locate any of my sections

2004-12-13 Thread Ed Greenberg

I downloaded phpconfig and set it up to read my config files, but it never 
successfully recognizes any of my sections. The regular expression seems to 
be included in the line:

if(preg_match(/^\s*\[([^\]]*)\].*[\r\n]\$/, $line))
and later, the same regex.
I'm not sure about the [\r\n] on the end of the line. My copy of the regex 
coach does not let me match a typical section header against it.

Could somebody (one of the authors, perhaps?) comment on this regex and on 
any other reasons why this code doesn't recognize ANY of my section headers 
in brackets?

I've looked at all the simple things My config files are very basic 
unix text files.

Thanks,
/edg
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[Asterisk-Users] CPU spikes with wcfxs loaded

2004-12-13 Thread Michael Welter

I need to reopen this discussion because it's impossible to run spandsp 
(and VoIP) under these circumstances.

With zaptel unloaded, I see the following vmstat 1 output:
no swapping, an occasional disk output, +/- 1003 interrupts/sec., less 
than 10 context switches/sec., CPU idle 100%.  A very quiet system.

I load modules zaptel and wcfxo, and the system utilization stays the 
same.  When I load wcfxs, the number of interrupts goes up to +-2004, 
which is normal.  However, every three seconds the CPU spikes to 50%. 
This is system utilization, not userland.  I assume it's in a wcfxs 
interrupt.

The number of interrupts stays constant at about 2004 during each spike, 
leading me to the conclusion that the TDM card is holding an interrupt 
for 500ms every three seconds (50% of 1000ms is 500ms).  This is a 
disaster for spandsp and VoIP in general.

When I unload the wcfxs module, CPU idle goes back to a constant 100%.
The TDM22B card is REV E/F, and I've tried it with several different 
cards.  Fedora Core 3 with linux-2.6.9 downloaded from kernel.org (a 
stock kernel).  The CPU is Athlon K7.

Can anyone please give me a clue?
Thanks,
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
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RE: [Asterisk-Users] Re: Dialing out to 2 clients simultaneously

2004-12-13 Thread niels

SER I am not happy with it.. didn't manage to get these below things
functioning:

It's too strict with authentication (user has to set specific
domain/realm) have problems with several types of hardphones authing to
SER

You can't make config changes without having to restart SER

Can't change the from-URI (CLID)

Can change to-URI (add prefix or something) but when you do that ser
behaves strange, in statefull mode not recognizing the packets that
follow (call leg does not exist etc) 

And some other minor things

I just had too many issues with it... 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Ivar
Helbekkmo
Sent: maandag 13 december 2004 18:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Dialing out to 2 clients simultaneously

[EMAIL PROTECTED] writes:

 Because else people will complain that they can't register two
 softphones anymore with same user/pass (because only one of the two
 softphones can receive the incoming calls) :-)

That's not what I meant -- that bit was clear.  I was wondering why it
is important to you to get rid of SER.  It is quite common to use SER
in front of Asterisk, and let each do what it does best.

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
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RE: [Asterisk-Users] music on hold garbled

2004-12-13 Thread Chris Cherry

Check your mpg123 make sure its not just a Symlink to mpg321

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Austad
Sent: Monday, December 13, 2004 9:55 AM
To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] music on hold garbled

Only tried it on X-lite, SIP, with ulaw and alaw.

On Dec 13, 2004, at 3:57 AM, Wilson Pickett wrote:

 It's very loud and has a lot of garbling in it.

 What/how many phones have you tried it on? What channels
 (ZAP/SIP/IAX2) and what codecs?
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Re: [Asterisk-Users] How to connect * to Adtran 600?

2004-12-13 Thread Gregory Junker

http://www.voip-info.org/wiki-Asterisk+Channel+Bank
Digium T100P, T1 cable to Adtran T1 port, extensions to Adtran FXS 
interfaces. Follow the instructions on the Wiki for configuring the 
T100P both in /etc/asterisk/zapata.conf and /etc/zaptel.conf. Configure 
the ports on the Adtran per the Adtran manual and you should be off and 
running.

Greg
Robert Augustyn wrote:
Hi,
I have been looking on that unit to be used as source of fxs ports.
Now I am not sure how I can get * box talking to it?
Thanks for advice.
robert

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RE: [Asterisk-Users] How to connect * to Adtran 600?

2004-12-13 Thread Shoval Tomer

You can use a T1 from digium for that
(http://www.digium.com/index.php?menu=wildcard_t100p). 

Please post using plain text next time.

From: Robert Augustyn [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, December 14, 2004 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to connect * to Adtran 600?

Hi,
I have been looking on that unit to be used as source of fxs ports.
Now I am not sure how I can get * box talking to it?
Thanks for advice.
robert

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RE: [Asterisk-Users] Dialing out to 2 clients simultaneously

2004-12-13 Thread Anders F Eriksson

Hi,

I don't think any SIP server would allow you to register more than once with
the same login information. What you can do in asterisk is setup two
different entries in sip.conf and then use extensions.conf to dial both.

Example from extensions.conf

[default]
exten = 1000,1,Dial(SIP/user1SIP/user2,60,t)
exten = 1000,2,Congestion
exten = 1000,3,Hangup
exten = 1000,102,Busy

/Anders

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 [EMAIL PROTECTED]
 Sent: den 13 december 2004 14:39
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Dialing out to 2 clients simultaneously
 
 Hmmm that's bad... 
 
 This is the last issue I have which makes that I can't get 
 rid of the SER proxy in front of asterisk.. Want to get rid of it
 
 Are there any plans to change this design?? (that multiple 
 UA's can register to one peer?)
 
 Niels

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Re: [Asterisk-Users] recommended IP phones and VoIP providers?

At 04:59 PM 12/13/04, you wrote:
Can anyone give me some recommendations for IP phones that work well with 
Asterisk?

I'm hoping for something not much more then $100 bux or so.
grandstream bt100 will work  100

Also does vonage service work directly through Asterisk or would I have to 
use their hardware? Or are there any other suggestions for a VoIP

provider?
vonage requires you to have their device and you need to have an fxo
of some sort to work with them, (from what their tech support told me).
They are not a pure voip provider - ethernet only requirement.
livevoip.com is one and there are others mentioned on voip-info.org
Greg
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[Asterisk-Users] The correct way to get most recent stable