Re: [asterisk-users] asterisk-users Digest, Vol 99, Issue 9

2012-10-05 Thread frangky robert

  Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom 
  PSTN gateway - pstn line to telcoi'm using xlite for windows
 
  when I make a phone call (sip - outgoing channel),I can hear my own voice 
  so clear. it's very annoying mewhen talking a little loud... any solution? 
 
 Two questions:
 
 (1) Does the problem occur when you make a SIP-to-SIP call, without
 the PSTN being involved?

No, it's happened only when I make a call from sip to pstn line.

 (2) When you hear your own voice in the headset, is it delayed, or
 is just an immediate louder-than-you-want side-tone?

it's immediate voice and very clear, just like talk-to-my-ear with no delay
 If it *does* occur in SIP-to-SIP calls, this would rule out your
 XORCOM and the PSTN as the cause.  If it's only occurring in
 SIP-to-PSTN calls, then the XORCOM and PSTN (or the interaction
 between them) is a likely suspect.
 
 There are several things which can cause this sort of problem.
 
 (A) Direct acoustic feedback within the headset.  In this case, you'd
 probably hear it even if the headset was unplugged entirely.  The
 only cure is to buy a better headset.
 
 (B) Incorrect audio-mixer settings in your PC.  To the PC audio
 infrastructure, a headset usually looks like a microphone
 and a separate speaker.  The audio mixer (hardware and software)
 usually has an ability to mix some of what the microphone hears
 into the speaker output.  If this knob is turned up too high,
 you'll hear your own voice too loudly.  If too low, you won't
 hear your own voice at all when you speak into the headset, and
 many people find this lack of side-tone to be confusing.
 
 The cure here is to adjust the audio side-tone level, either
 in your Windows audio-mixer control panel, or in X-Lite (if
 it has such an adjustment).
 
 (C) Electrical reflection from an analog impedance discontinuity
 in the analog telephone-line system.  This can result from
 a mismatch between the telephone wiring, and the PSTN interface
 device, and can occur at any point in the analog transmission.
 
 If the loud side-tone you hear is *not* delayed noticeably,
 then the impedance mismatch might be at your XORCOM/PSTN
 interface.  The XORCOM may have a software adjustment or
 jumper setting, to match its audio impedance to that of your
 local phone line... try fiddling with these settings to see
 if they reduce the excessive side-tone level.
 
 If the loud side-tone you hear is delayed (it sounds a bit
 like an echo) then it may very well be at the far end of
 the phone line, outside of your own physical control... it
 might be at your local phone office, or anywhere between you
 and the far end of the phone connection.  Not much you can do
 about this.
 
 (D) Audio feedback at the far end of the call, in a cheap phone
 handset.  Sometimes, audio from the back side of the speaker
 in a handset travels through the body of the handset and is
 picked up by the microphone, and results in an audible delayed
 echo of the voice from the far end of the line.  Using a
 better handset, or stuffing the handset full of audio damping
 material (cloth or cotton or fiberglass) is the cure here.

Well, thanks a lot Lee for suggestion and explanation, I'll try this tommorow.

 
 We've often faced this problem with SIP soft phones when the computer's 
 sound system gain was set too high.  You usually have to play around 
 with microphone gain settings to get to the point where the echo 
 disappears with the other party still being able to hear you.

And thanks for your share Raj, I appreciate that..  
  --
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] I can hear my own voice through the headset

2012-10-05 Thread frangky robert


Sorry for my last post,




  Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom 
  PSTN gateway - pstn line to telcoi'm using xlite for windows
 
  when I make a phone call (sip - outgoing channel),I can hear my own voice 
  so clear. it's very annoying mewhen talking a little loud... any solution? 
 
 Two questions:
 
 (1) Does the problem occur when you make a SIP-to-SIP call, without
 the PSTN being involved?

No, it's happened only when I make a call from sip to pstn line.

 (2) When you hear your own voice in the headset, is it delayed, or
 is just an immediate louder-than-you-want side-tone?

it's immediate voice and very clear, just like talk-to-my-ear with no delay
 If it *does* occur in SIP-to-SIP calls, this would rule out your
 XORCOM and the PSTN as the cause.  If it's only occurring in
 SIP-to-PSTN calls, then the XORCOM and PSTN (or the interaction
 between them) is a likely suspect.
 
 There are several things which can cause this sort of problem.
 
 (A) Direct acoustic feedback within the headset.  In this case, you'd
 probably hear it even if the headset was unplugged entirely.  The
 only cure is to buy a better headset.
 
 (B) Incorrect audio-mixer settings in your PC.  To the PC audio
 infrastructure, a headset usually looks like a microphone
 and a separate speaker.  The audio mixer (hardware and software)
 usually has an ability to mix some of what the microphone hears
 into the speaker output.  If this knob is turned up too high,
 you'll hear your own voice too loudly.  If too low, you won't
 hear your own voice at all when you speak into the headset, and
 many people find this lack of side-tone to be confusing.
 
 The cure here is to adjust the audio side-tone level, either
 in your Windows audio-mixer control panel, or in X-Lite (if
 it has such an adjustment).
 
 (C) Electrical reflection from an analog impedance discontinuity
 in the analog telephone-line system.  This can result from
 a mismatch between the telephone wiring, and the PSTN interface
 device, and can occur at any point in the analog transmission.
 
 If the loud side-tone you hear is *not* delayed noticeably,
 then the impedance mismatch might be at your XORCOM/PSTN
 interface.  The XORCOM may have a software adjustment or
 jumper setting, to match its audio impedance to that of your
 local phone line... try fiddling with these settings to see
 if they reduce the excessive side-tone level.
 
 If the loud side-tone you hear is delayed (it sounds a bit
 like an echo) then it may very well be at the far end of
 the phone line, outside of your own physical control... it
 might be at your local phone office, or anywhere between you
 and the far end of the phone connection.  Not much you can do
 about this.
 
 (D) Audio feedback at the far end of the call, in a cheap phone
 handset.  Sometimes, audio from the back side of the speaker
 in a handset travels through the body of the handset and is
 picked up by the microphone, and results in an audible delayed
 echo of the voice from the far end of the line.  Using a
 better handset, or stuffing the handset full of audio damping
 material (cloth or cotton or fiberglass) is the cure here.

Well, thanks a lot Lee for suggestion and explanation, I'll try this tommorow.

 
 We've often faced this problem with SIP soft phones when the computer's 
 sound system gain was set too high.  You usually have to play around 
 with microphone gain settings to get to the point where the echo 
 disappears with the other party still being able to hear you.

And thanks for your share Raj, I appreciate that..  
  --
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] I can hear my own voice through the headset

2012-10-03 Thread frangky robert




Hi all,
Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom 
PSTN gateway - pstn line to telcoi'm using xlite for windows
when I make a phone call (sip - outgoing channel),I can hear my own voice so 
clear. it's very annoying mewhen talking a little loud... any solution? 
Thanks,
Frangky
  --
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] CallerID problem with astribank

2010-05-03 Thread frangky robert

Hi all... I'm sorry for repeating my message.
 
I have a problem with caller id on my asterisk server with xorcom astribank.

here is my configuration :
centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2
ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting)
 
everything fine until I try to feed my app with caller id.

 
here is the log :
 
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:24:40] ERROR[30895]: callerid.c:562 callerid_feed: No start bit 
found in fsk data.
[Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8712 ss_thread: CallerID feed 
failed: Success
[Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8816 ss_thread: CallerID 
returned with error on channel 'DAHDI/15-1'

 
 
Please help me 

  
_
NEW! Get Windows Live FREE.
http://www.get.live.com/wl/all-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Caller ID on Asterisk and Astribank

2010-04-29 Thread frangky robert

Hi all...

I have a problem with caller id on my asterisk server.
here is my configuration :
centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2
ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting)

everything fine until I try to feed my app with caller id.

My extensions.conf :

[incoming1]
exten = s,1,AGI(/var/apps/core/runagi,incoming,${CALLERID(num)})
exten = s,n,QUEUE(${que},trkd)
exten = h,1,Hangup()

here is the log :

-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:24:40] ERROR[30895]: callerid.c:562 callerid_feed: No start bit 
found in fsk data.
[Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8712 ss_thread: CallerID feed 
failed: Success
[Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8816 ss_thread: CallerID 
returned with error on channel 'DAHDI/15-1'
  == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
  == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to 
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new 
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1, 
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi
-- Playing 'en/0006' (escape_digits=) (sample_offset 0)

I read instructions from a few forums
then I made a change on 'chan_dahdi.conf' like :
-
1:
cidsignalling=v23, cidstart=ring, hidecallerid=no, callerid=asreceived

Here's the log :

-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:42:03] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 18 
(Ring Begin)...
[Apr 30 11:42:03] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 2 
(Ring/Answered)...
[Apr 30 11:42:05] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 18 
(Ring Begin)...
  == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
  == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to 
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
[Apr 30 11:42:06] WARNING[31296]: chan_dahdi.c:6174 dahdi_handle_event: 
Ring/Off-hook in strange state 6 on channel 15
  == Spawn extension (default, s, 2) exited non-zero on 'DAHDI/15-1'


-
2:

cidsignalling=dtmf', cidstart=ring, hidecallerid=no, callerid=asreceived

Here's the log :

-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:49:28] WARNING[31491]: chan_dahdi.c:8610 ss_thread: DTMFCID timed 
out waiting for ring. Exiting simple switch
-- Hungup 'DAHDI/15-1'
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:49:34] DEBUG[31492]: chan_dahdi.c:8630 ss_thread: CID is '', flags 8
  == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
  == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to 
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
[Apr 30 11:49:35] WARNING[31492]: chan_dahdi.c:6174 dahdi_handle_event: 
Ring/Off-hook in strange state 6 on channel 15
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new 
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1, 
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi


-
3:



cidsignalling=dtmf', cidstart=polarity, hidecallerid=no, callerid=asreceived



Here's the log :

   -- Starting simple switch on 'DAHDI/15-1'
  == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
  == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to 
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
-- Registered IAX2 '9009' (AUTHENTICATED) at 127.0.0.1:48961
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new 
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1, 
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi


-
4:





cidsignalling=v23', cidstart=polarity, hidecallerid=no, 
callerid=asreceived





Here's the log :


-- Starting simple switch on 'DAHDI/15-1'
  == Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
  == Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to 
context 'default'
-- Executing [...@default:1] 

Re: [asterisk-users] Astribank problem

2010-02-03 Thread frangky robert

Ok, the problem solved...

thanks for your advice.

after rebooting i run /usr/share/dahdi/xpp_fxloader load
and everything run normally.

thanks...

 Message: 24
 Date: Mon, 1 Feb 2010 11:03:44 +0200
 From: Tzafrir Cohen tzafrir.co...@xorcom.com
 Subject: Re: [asterisk-users] Astribank problem
 To: asterisk-users@lists.digium.com
 Message-ID: 20100201090344.gr3...@xorcom.com
 Content-Type: text/plain; charset=us-ascii
 
 On Mon, Feb 01, 2010 at 07:42:51AM +, frangky robert wrote:
  
  
  
  
  I do some test:
  1.unplug usb connector from server to astricon
  2.unplug power to astricon
  3.plug-in the power to astricon
  4.plug-in the usb connector
  
  Here is the log from /var/log/messages after doing the 1st step.
  
  Feb  1 19:38:24 localhost last message repeated 2 times
  Feb  1 19:43:39 localhost kernel: ERR-xpp_usb: xusb-0 (usb-:00:1d.7-3) 
  [X1038295]: nonzero write bulk status received: -71 (pending_writes=1)
  Feb  1 19:43:39 localhost kernel: usb 2-3: USB disconnect, address 3
  Feb  1 19:43:39 localhost kernel: ERR-xpp_usb: XBUS-00: xusb_listen: 
  usb_submit_urb failed: -19
  Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] 
  Disconnecting
  Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] 
  Deactivating
  Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Release 
  XPDS
  Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-00: Remove
  Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-10: Remove
  Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-20: Remove
  Feb  1 19:43:39 localhost kernel: NOTICE-xpp: worker_reset: 
  worker(XBUS-00)-xpds_init_done=0
  Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] 
  Atribank Remove
  Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] 
  Astribank Release
  Feb  1 19:43:39 localhost kernel: INFO-xpp_usb: xusb-0 (usb-:00:1d.7-3) 
  [X1038295]: now disconnected
  Feb  1 19:43:39 localhost 'astribank_hook'[3728]: offline(XBUS-00): 0/1 
  from /etc/dahdi/xpp_order
  Feb  1 19:43:39 localhost 'astribank_hook'[3735]: All Astribanks offline
  
  
  And, this is the log after doing 4th step.
  
  Feb  1 19:44:20 localhost kernel: usb 2-3: new high speed USB device using 
  ehci_hcd and address 4
  Feb  1 19:44:20 localhost kernel: usb 2-3: configuration #1 chosen from 1 
  choice
  Feb  1 19:44:21 localhost 'xpp_fxloader'[3847]: Exiting... 
  XPP_HOTPLUG_DISABLED
 
 Seems like you explicitly disabled firmware loading by setting
 XPP_HOTPLUG_DISABLED in /etc/dahdi/init.conf . Just rem-out that line.
 
  
  lsusb result is:
  
  [r...@localhost ~]# lsusb
  Bus 002 Device 004: ID e4e4:1160
  Bus 002 Device 001: ID :
  Bus 006 Device 001: ID :
  Bus 006 Device 002: ID 04b3:3025 IBM Corp.
  Bus 004 Device 001: ID :
  Bus 008 Device 001: ID :
  Bus 007 Device 001: ID :
  Bus 001 Device 001: ID :
  Bus 005 Device 001: ID :
  Bus 003 Device 001: ID :
  
  here is the msg when i do /usr/share/dahdi/xpp_fxloader
  [r...@localhost ~]# /usr/share/dahdi/xpp_fxloader usb
 
 This only runs USB firmware loading. And as the firmware loading is
 explicitly disabled on your system, the FPGA firmware will still not get
 loaded.
 
 This is also something that you would have seen if you would run
 'dahdi_hardware -v'
 
 So basically just remove that line from init.conf and replug the
 Astribank.
 
 -- 
Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
 
 
 
 *
  
_
New Windows 7: Find the right PC for you. Learn more.
http://windows.microsoft.com/shop-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Astribank problem

2010-01-31 Thread frangky robert




I do some test:
1.unplug usb connector from server to astricon
2.unplug power to astricon
3.plug-in the power to astricon
4.plug-in the usb connector

Here is the log from /var/log/messages after doing the 1st step.

Feb  1 19:38:24 localhost last message repeated 2 times
Feb  1 19:43:39 localhost kernel: ERR-xpp_usb: xusb-0 (usb-:00:1d.7-3) 
[X1038295]: nonzero write bulk status received: -71 (pending_writes=1)
Feb  1 19:43:39 localhost kernel: usb 2-3: USB disconnect, address 3
Feb  1 19:43:39 localhost kernel: ERR-xpp_usb: XBUS-00: xusb_listen: 
usb_submit_urb failed: -19
Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] 
Disconnecting
Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Deactivating
Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Release XPDS
Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-00: Remove
Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-10: Remove
Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00/XPD-20: Remove
Feb  1 19:43:39 localhost kernel: NOTICE-xpp: worker_reset: 
worker(XBUS-00)-xpds_init_done=0
Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Atribank 
Remove
Feb  1 19:43:39 localhost kernel: INFO-xpp: XBUS-00: [usb:X1038295] Astribank 
Release
Feb  1 19:43:39 localhost kernel: INFO-xpp_usb: xusb-0 (usb-:00:1d.7-3) 
[X1038295]: now disconnected
Feb  1 19:43:39 localhost 'astribank_hook'[3728]: offline(XBUS-00): 0/1 from 
/etc/dahdi/xpp_order
Feb  1 19:43:39 localhost 'astribank_hook'[3735]: All Astribanks offline


And, this is the log after doing 4th step.

Feb  1 19:44:20 localhost kernel: usb 2-3: new high speed USB device using 
ehci_hcd and address 4
Feb  1 19:44:20 localhost kernel: usb 2-3: configuration #1 chosen from 1 choice
Feb  1 19:44:21 localhost 'xpp_fxloader'[3847]: Exiting... XPP_HOTPLUG_DISABLED

lsusb result is:

[r...@localhost ~]# lsusb
Bus 002 Device 004: ID e4e4:1160
Bus 002 Device 001: ID :
Bus 006 Device 001: ID :
Bus 006 Device 002: ID 04b3:3025 IBM Corp.
Bus 004 Device 001: ID :
Bus 008 Device 001: ID :
Bus 007 Device 001: ID :
Bus 001 Device 001: ID :
Bus 005 Device 001: ID :
Bus 003 Device 001: ID :

here is the msg when i do /usr/share/dahdi/xpp_fxloader
[r...@localhost ~]# /usr/share/dahdi/xpp_fxloader usb
'xpp_fxloader'[3955]: - FIRMWARE LOADING: (usb) [1 devices]
Got all 1 devices
'xpp_fxloader'[4074]: - FIRMWARE IS LOADED

but, when i do /etc/init.d/dahdi stop and start
here is the result

[r...@localhost ~]# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
  xpp_usb: [  OK  ]

Astribanks initialization is starting
Astribanks detection ..TIMEOUT
No hardware timing source found in /proc/dahdi, loading dahdi_dummy
Running dahdi_cfg:  DAHDI_CHANCONFIG failed on channel 1: No such device or 
address (6)
   [FAILED]


On Sat, Jan 30, 2010 at 03:57:30AM +, frangky robert wrote:
 
 H all...
 
 I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, 
 dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final.
 My problem is, every time i unplug the astribank power supply, and 
 reconnect it, astribank cannot work again (lsusb result is 11x0)... 
 
When you plug the Astribank, the firmware should get loaded by a script
called from udev.
 
What messages do you see in /var/log/messages following that?
 
 but, after reinstall the asterisk and dahdi, astribank will detected
 (lsusb result is 11x2)... 
 
-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
  
_
Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail 
you.
http://www.microsoft.com/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-id:SI_SB_3:092010-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Astribank problem

2010-01-29 Thread frangky robert

H all...

I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, 
dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final.
My problem is, every time i unplug the astribank power supply, and 
reconnect it, astribank cannot work again (lsusb result is 11x0)... 
but, after reinstall the asterisk and dahdi, astribank will detected (lsusb 
result is 11x2)... 

any suggestion? 

Regard,


frank.
  
_
Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail 
you.
http://www.microsoft.com/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-id:SI_SB_3:092010-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] memory leak on asterisk 1.6.0.6

2009-05-23 Thread frangky robert

for the second time i'm asking in this forum, 
somebody help me

my asterisk box have a problem with memory leak.
I'm scheduling to rstart the box to fix this problem
but any cleverer suggest to fix this? coz this issue
causing another problem to my AGI application...


thankyou before

_
NEW! Get Windows Live FREE.
http://www.get.live.com/wl/all___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Memory leak on asterisk 1.6.0.6

2009-05-22 Thread frangky robert


Hi everyones,

I have a production server using asterisk 1.6.0.6
using php as an IVR and mssql server (on other machine)
My server attached a Sangoma A104 card (4xT1 card)

i have a problem with memory leak on that server
and causing a delay on IVR prompt. (Thats my assumption, memory leak problem)

any suggestion on this problem? or maybe any assumption?



frangky

_
Join the Fantasy Football club and win cash prizes here!
http://fantasyfootball.id.msn.com___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users