date:20070321

Hi,

At one location we have a user whose Polycom IP430 suffers from erratic 
reboots. We swapped his phone for a brand new one, but the problem 
remains.

Has anyone seen that?
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Re: [asterisk-users] Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation

2007-03-21 Thread dave cantera





george,
is this a production system you are upgrading to 1.4?
daveC

George C. Attopany wrote:

Hi,
  
  
I run ASTERISK 1.2 with a Wildcard TE410P-Xilinx on Redhat Linux
8.0(Psyche) with Kernel 2.4.18-14 on an i686 with a Channel bank to
enable me use anlog handsets.
  
  
I would like to upgrade to Fedora Core 6 and then run ASTERISK 1.4.1
and latest Zaptel etc.
  
  
Reading from the mailing list about problems with Fedora and Linux
Kernel 2.6, and from the link (
http://www.voip-info.org/wiki/index.php?page=Asterisk+OS+Platforms ) I
am a bit confused about the way forward with my upgrade.
  
  
I am not very good with Linux systems, but would appreciate your advice
to sail through my upgrade successful. Any hint from you would be
welcome, in particular how to upgrade my platform to Fedora Core 6 and
the necessary Kernel to give me a stable platform.
  
  
Looking forward to hearing from you.
  
  
  
Kind regards,
  
  
george.
  
  
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RE: [asterisk-users] Problem with ATT Maintenance protocol in PRIconnection, no B+D channels available

2007-03-21 Thread Kanelbullar

I meant the 24th, yes. Could there be any problem with the support to the ATT 
maintenance protocol? We have checked the libpri code, in q931.c, and there 
appears to be a simple response to messages that have the protocol 
discriminator set to ATT maintenance: a copy with a changed byte.
   
  if ((h-pd == 0x3) || (h-pd == 0x43)) {
/* This is the weird maintenance stuff.  We majorly
   KLUDGE this by changing byte 4 from a 0xf (SERVICE)
   to a 0x7 (SERVICE ACKNOWLEDGE) */
h-raw[h-crlen + 2] -= 0x8;
q931_xmit(pri, h, len, 1);
return 0;
}

  Could there be cases where a more specific response might be needed?
   
  
Michael Collins [EMAIL PROTECTED] escreveu:
   I've never seen a PRI dchannel on a T1 on a timeslot other than the
 24th. Are you sure that it's really on channel 23?

I think he meant channel 23 of channels 0~23, aka the 24th channel.
-MC


 
 Matthew Fredrickson
 
 On Mar 20, 2007, at 8:54 AM, Kanelbullar wrote:
 
  Thanks for your answer, Bruno. However, the configuration you provided
  is for an E1 connection and we are using a T1, having channel 23 as D
  channel.
 
  Bruno De Luca escreveu:d-channel is in midle
 
  bchan=1-15,17-31
  dchan=16
  loadzone = it
  defaultzone = it
 
 
 
 
  Kanelbullar wrote:Hi guys,
 
  We are experiencing a problem with a T1 PRI connection. After trying
  a number of variations in the configuration files, the behavior is
  always the same: no B channels come up and the D channel doesn't
  appear to be working well. We can see there are ATT Maintenance
  messages being exchanged by asterisk and the provider, CONNECT and
  CONNECT ACKNOWLEDGE, but that doesn't appear to be enough to bring
  the D and B channels properly up. Are there any messages missing?
  When we attempt to make a call, we can see the Q.931 SETUP message
  being sent. But shortly after we are getting a LAPD DISC message,
  which ends up originating a Q.931 DISCONNECT message, terminating
  the call.
 
  What could be the problem here?
  * Could there be any configuration issue on our side?
  * Does libpri provide complete support to the ATT Maintenance
  protocol or could this connection be incompatible?
 
  Any help would be highly appreciated.
 
  Many thanks in advance,
  Paulo
 
  
  PS: Configuration files, messages and pri debug snippets follow
 
  zaptel.conf
  
  loadzone = us
  defaultzone=us
  #Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1  PRI_T1
  span=1,0,0,esf,b8zs,crc4
  bchan=1-23
  dchan=24
 
  zapata.conf
  
  [channels]
  group = 0
  usecallingpres = yes
  switchtype = national
  context = inbound
  signalling = pri_cpe
  usecallerid = yes
  channel = 1-23
 
  messages
  --
  Mar 19 15:32:23 NOTICE[3306] cdr.c: CDR logging disabled, data will
  be lost.
  Mar 19 15:32:23 WARNING[3306] pbx_ael.c: Unable to open
  '/etc/asterisk/extensions.ael': No such file or directory
  Mar 19 15:32:23 WARNING[3306] pbx.c: Requested contexts didn't get
  merged
  Mar 19 15:33:17 WARNING[3322] chan_zap.c: No D-channels available!
  Using Primary channel 24 as D-channel anyway!
  Mar 19 15:33:58 WARNING[3322] chan_zap.c: No D-channels available!
  Using Primary channel 24 as D-channel anyway!
  Mar 19 15:33:58 WARNING[3366] app_dial.c: Unable to forward voice
  [...]
 
  pri debug span
  --
   [ 00 01 0a 0a 03 01 00 07 01 01 c0 18 01 ac ]
   Informational frame:
   SAPI: 00  C/R: 0 EA: 0
TEI: 000EA: 1
   N(S): 005   0: 0
   N(R): 005   P: 0
   10 bytes of data
  -- Restarting T203 counter
  Stopping T_203 timer
  Starting T_200 timer
   Protocol Discriminator: ATT Maintenance (3)  len=10
   Call Ref: len= 1 (reference 0/0x0) (Originator)
   Message type: CONNECT (7)
   [01 01 c0]
   IE: Change Status (len = 3)
   [18 01 ac]
   Channel ID (len= 3) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
  Exclusive Dchan: 1
  ChanSel: As indicated in following octets
   ]
  (...)
   [ 02 01 0a 0c 03 01 00 0f 01 01 c0 18 01 ac ]
   Informational frame:
   SAPI: 00  C/R: 1 EA: 0
TEI: 000EA: 1
   N(S): 005   0: 0
   N(R): 006   P: 0
   10 bytes of data
  -- ACKing all packets from 5 to (but not including) 6
  -- Since there was nothing left, stopping T200 counter
  -- Stopping T203 counter since we got an ACK
  -- Nothing left, starting T203 counter
   Protocol Discriminator: ATT Maintenance (3)  len=10
   Call Ref: len= 1 (reference 0/0x0) (Originator)
   Message type: CONNECT ACKNOWLEDGE (15)
   [01 01 c0]
   IE: Change Status (len = 3)
   [18 01 ac]
   Channel ID (len= 3) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
  Exclusive Dchan: 1
  ChanSel: As indicated in following octets
   ]
  (...)
   Protocol Discriminator: Q.931 (8)  len=40
   Call Ref: len= 2

RE: [asterisk-users] Problem with ATT Maintenance protocol inPRI connection, no B+D channels available

2007-03-21 Thread Kanelbullar

Thank you guys. We have tried both ways, with and without crc4, and the result 
is quite the same. There appears to be a problem we the specific connection we 
are using and the ATT Maintenance stuff.

Michael Collins [EMAIL PROTECTED] escreveu:span=1,0,0,esf,b8zs,crc4
 
 This needs to be span=1,1,0,esf,b8zs
 
 I'm not sure if the crc4 is necessary.
 
 Doug

I concur with Doug. I have two PRI's in one system. My zaptel.conf
looks like this:

span=1,1,0,esf,b8zs # PRI line - LD Qwest (interstate)
bchan=1-23
dchan=24
span=2,2,0,esf,b8zs # PRI line - SBCLD (intrastate/local)
bchan=25-47
dchan=48

HTH,
MC
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Re: [asterisk-users] polycom random reboots

2007-03-21 Thread Bruce Reeves


Yes, I recently saw this with a 501, in my case the network drop was
the problem. If you have a good tester then run it on the connection.
I had another drop near by and just swicthed to it.

On 3/21/07, Louis-David Mitterrand
[EMAIL PROTECTED] wrote:

Hi,

At one location we have a user whose Polycom IP430 suffers from erratic
reboots. We swapped his phone for a brand new one, but the problem
remains.

Has anyone seen that?
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--
Bruce
Nortex Networks
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Re: [asterisk-users] polycom random reboots

2007-03-21 Thread joe a.

Did you swap the power module as well?  If POE, did you swap the patch cord?

If the power module plugs into a power strip did you change that? or at least 
the position in the strip?

joe a.

Louis-David Mitterrand[EMAIL PROTECTED] Wrote: 3/21/2007 6:40 AM:
 Hi,
 
 At one location we have a user whose Polycom IP430 suffers from erratic 
 reboots. We swapped his phone for a brand new one, but the problem 
 remains.
 
 Has anyone seen that?
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Re: [asterisk-users] polycom random reboots

2007-03-21 Thread Henry Cobb


On 3/21/07, Louis-David Mitterrand
[EMAIL PROTECTED] wrote:

Hi,

At one location we have a user whose Polycom IP430 suffers from erratic
reboots. We swapped his phone for a brand new one, but the problem
remains.

Has anyone seen that?


Our Polycom 3s and 5s ship with flaky power supplies and tend to
reboot all of the time (especially in India...), so we found
replacement non-Polycom power supplies and they are much more stable.

-HJC
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Re: [asterisk-users] polycom random reboots

On Wed, Mar 21, 2007 at 07:07:00AM -0400, joe a. wrote:
 Did you swap the power module as well?  If POE, did you swap the 
 patch cord?
 
 If the power module plugs into a power strip did you change that? or 
 at least the position in the strip?

Thanks for the tought, but the IP430 has no external power strip or 
module, it's fully integrated like the IP601.

We changed the cable, the wall socket and the switch (was due for an 
upgrade). Now on to testing the LAN.
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Re: [asterisk-users] polycom random reboots

On Wed, Mar 21, 2007 at 04:07:34AM -0700, Henry Cobb wrote:
 On 3/21/07, Louis-David Mitterrand
 [EMAIL PROTECTED] wrote:
 Hi,
 
 At one location we have a user whose Polycom IP430 suffers from erratic
 reboots. We swapped his phone for a brand new one, but the problem
 remains.
 
 Has anyone seen that?
 
 Our Polycom 3s and 5s ship with flaky power supplies and tend to
 reboot all of the time (especially in India...), so we found
 replacement non-Polycom power supplies and they are much more stable.

I should have added that we use POE with a 3com PWR-class switch.
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Re: [asterisk-users] polycom random reboots

On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote:
 Yes, I recently saw this with a 501, in my case the network drop was
 the problem. If you have a good tester then run it on the connection.
 I had another drop near by and just swicthed to it.

What kind of test tool would you suggest? Usually we rely on the cabling 
guys for that but that entails a delay and I'd be interested in knowing 
how to do it myself.

Thanks,
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Re: [asterisk-users] Tesco Internet Phone

On Friday 02 March 2007 07:46, Alan Chandler wrote:

 On Thursday 01 March 2007 20:33, bails wrote:
  plug it in a linux box and tell us what it is please,
  generic-usb-audio or what?
 
  Bails
 
  Julian Lyndon-Smith wrote:
   Yeah, that's where firefly comes from, doesn't it.
  
   I've got the base station plugged in, and the handset connected to
   it, but it always says pc unavailable.
  
   My system (xp) sees a usb phone for speakers and microphone,
   but I can't get it to work.

 Did this go any further.  I would be interested in this.

I too would really like to find or help adapt a driver for this.

Here's what I get from my USB-DECT device:

# cat /proc/bus/usb/devices
T:  Bus=01 Lev=02 Prnt=02 Port=00 Cnt=01 Dev#=  6 Spd=12  MxCh= 0
D:  Ver= 1.10 Cls=00(ifc ) Sub=00 Prot=00 MxPS=64 #Cfgs=  1
P:  Vendor=19af ProdID=694d Rev= 0.00
S:  Manufacturer=innoMax Technology Ltd.
S:  Product=Cordless USB Phone
C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=400mA
I:  If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=audio
I:  If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 1 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=83(I) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=02(O) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 3 Alt= 0 #EPs= 2 Cls=03(HID  ) Sub=00 Prot=00 Driver=hid
E:  Ad=81(I) Atr=03(Int.) MxPS=  32 Ivl=1ms
E:  Ad=01(O) Atr=03(Int.) MxPS=  32 Ivl=1ms

# lsusb -v -s 006

Bus 001 Device 006: ID 19af:694d
Device Descriptor:
  bLength18
  bDescriptorType 1
  bcdUSB   1.10
  bDeviceClass0 (Defined at Interface level)
  bDeviceSubClass 0
  bDeviceProtocol 0
  bMaxPacketSize064
  idVendor   0x19af
  idProduct  0x694d
  bcdDevice0.00
  iManufacturer   1 innoMax Technology Ltd.
  iProduct2 Cordless USB Phone
  iSerial 0
  bNumConfigurations  1
  Configuration Descriptor:
bLength 9
bDescriptorType 2
wTotalLength  214
bNumInterfaces  4
bConfigurationValue 1
iConfiguration  0
bmAttributes 0x80
MaxPower  400mA
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber0
  bAlternateSetting   0
  bNumEndpoints   0
  bInterfaceClass 1 Audio
  bInterfaceSubClass  1 Control Device
  bInterfaceProtocol  0
  iInterface  0
  AudioControl Interface Descriptor:
bLength10
bDescriptorType36
bDescriptorSubtype  1 (HEADER)
bcdADC   1.00
wTotalLength   60
bInCollection   2
baInterfaceNr( 0)   1
baInterfaceNr( 1)   2
  AudioControl Interface Descriptor:
bLength12
bDescriptorType36
bDescriptorSubtype  2 (INPUT_TERMINAL)
bTerminalID 3
wTerminalType  0x0101 USB Streaming
bAssocTerminal  4
bNrChannels 1
wChannelConfig 0x
iChannelNames   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 9
bDescriptorType36
bDescriptorSubtype  3 (OUTPUT_TERMINAL)
bTerminalID 4
wTerminalType  0x0301 Speaker
bAssocTerminal  3
bSourceID   5
iTerminal   0
  AudioControl Interface Descriptor:
bLength12
bDescriptorType36
bDescriptorSubtype  2 (INPUT_TERMINAL)
bTerminalID 1
wTerminalType  0x0201 Microphone
bAssocTerminal  2
bNrChannels 1
wChannelConfig 0x
iChannelNames   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 9
bDescriptorType36
bDescriptorSubtype  3 (OUTPUT_TERMINAL)
bTerminalID 2
wTerminalType  0x0101 USB Streaming
bAssocTerminal  1
bSourceID   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 8
bDescriptorType36
bDescriptorSubtype  6 (FEATURE_UNIT)
bUnitID 5
bSourceID   3
bControlSize1
bmaControls( 0)  0x03
  Mute
  Volume
iFeature0
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber1

Re: [asterisk-users] polycom random reboots

2007-03-21 Thread Chris Mason (Lists)

The 501 are extremely sensitive to power fluctuations and will reboot as 
a result of a power transient even though every other piece of equipment 
is fine.


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(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
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[asterisk-users] Limit call duration

2007-03-21 Thread Suity Zsolt


Hi everyone,

I'm new to Asterisk, but I like it ;o)
Have a question to you;

How can I limit the incoming call duration?


--
Suich
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Re: [asterisk-users] Tesco Internet Phone

2007-03-21 Thread bails


Antony Stone wrote:

On Friday 02 March 2007 07:46, Alan Chandler wrote:


On Thursday 01 March 2007 20:33, bails wrote:

plug it in a linux box and tell us what it is please,
generic-usb-audio or what?

Bails

Julian Lyndon-Smith wrote:

Yeah, that's where firefly comes from, doesn't it.

I've got the base station plugged in, and the handset connected to
it, but it always says pc unavailable.

My system (xp) sees a usb phone for speakers and microphone,
but I can't get it to work.

Did this go any further.  I would be interested in this.


I too would really like to find or help adapt a driver for this.

Here's what I get from my USB-DECT device:

# cat /proc/bus/usb/devices
T:  Bus=01 Lev=02 Prnt=02 Port=00 Cnt=01 Dev#=  6 Spd=12  MxCh= 0
D:  Ver= 1.10 Cls=00(ifc ) Sub=00 Prot=00 MxPS=64 #Cfgs=  1
P:  Vendor=19af ProdID=694d Rev= 0.00
S:  Manufacturer=innoMax Technology Ltd.
S:  Product=Cordless USB Phone
C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=400mA
I:  If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=audio
I:  If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 1 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=83(I) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=02(O) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 3 Alt= 0 #EPs= 2 Cls=03(HID  ) Sub=00 Prot=00 Driver=hid
E:  Ad=81(I) Atr=03(Int.) MxPS=  32 Ivl=1ms
E:  Ad=01(O) Atr=03(Int.) MxPS=  32 Ivl=1ms

# lsusb -v -s 006

Bus 001 Device 006: ID 19af:694d
Device Descriptor:
  bLength18
  bDescriptorType 1
  bcdUSB   1.10
  bDeviceClass0 (Defined at Interface level)
  bDeviceSubClass 0
  bDeviceProtocol 0
  bMaxPacketSize064
  idVendor   0x19af
  idProduct  0x694d
  bcdDevice0.00
  iManufacturer   1 innoMax Technology Ltd.
  iProduct2 Cordless USB Phone
  iSerial 0
  bNumConfigurations  1
  Configuration Descriptor:
bLength 9
bDescriptorType 2
wTotalLength  214
bNumInterfaces  4
bConfigurationValue 1
iConfiguration  0
bmAttributes 0x80
MaxPower  400mA
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber0
  bAlternateSetting   0
  bNumEndpoints   0
  bInterfaceClass 1 Audio
  bInterfaceSubClass  1 Control Device
  bInterfaceProtocol  0
  iInterface  0
  AudioControl Interface Descriptor:
bLength10
bDescriptorType36
bDescriptorSubtype  1 (HEADER)
bcdADC   1.00
wTotalLength   60
bInCollection   2
baInterfaceNr( 0)   1
baInterfaceNr( 1)   2
  AudioControl Interface Descriptor:
bLength12
bDescriptorType36
bDescriptorSubtype  2 (INPUT_TERMINAL)
bTerminalID 3
wTerminalType  0x0101 USB Streaming
bAssocTerminal  4
bNrChannels 1
wChannelConfig 0x
iChannelNames   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 9
bDescriptorType36
bDescriptorSubtype  3 (OUTPUT_TERMINAL)
bTerminalID 4
wTerminalType  0x0301 Speaker
bAssocTerminal  3
bSourceID   5
iTerminal   0
  AudioControl Interface Descriptor:
bLength12
bDescriptorType36
bDescriptorSubtype  2 (INPUT_TERMINAL)
bTerminalID 1
wTerminalType  0x0201 Microphone
bAssocTerminal  2
bNrChannels 1
wChannelConfig 0x
iChannelNames   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 9
bDescriptorType36
bDescriptorSubtype  3 (OUTPUT_TERMINAL)
bTerminalID 2
wTerminalType  0x0101 USB Streaming
bAssocTerminal  1
bSourceID   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength 8
bDescriptorType36
bDescriptorSubtype  6 (FEATURE_UNIT)
bUnitID 5
bSourceID   3
bControlSize1
bmaControls( 0)  0x03
  Mute
  Volume
iFeature0
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber1

Re: [asterisk-users] Limit call duration

2007-03-21 Thread Robert Lister

On Wed, Mar 21, 2007 at 12:56:55PM +0100, Suity Zsolt wrote:
 Hi everyone,
 
 I'm new to Asterisk, but I like it ;o)
 Have a question to you;
 
 How can I limit the incoming call duration?

I think you can say something like:

AbsoluteTimeout (or in 1.2x, Set(TIMEOUT(absolute) = seconds) )

See: http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout

Rob

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Re: [asterisk-users] polycom random reboots

2007-03-21 Thread Bruce Reeves


The one I use the most is a Fluke Net Tool. It can determine polarity
problems and I believe has some diagnostics for POE and VOIP. It also can go
in-line between a device and the network and help diagnose problems the
device is experiencing that the tool would not encounter on its own.

On 3/21/07, Louis-David Mitterrand [EMAIL PROTECTED]
wrote:


On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote:
 Yes, I recently saw this with a 501, in my case the network drop was
 the problem. If you have a good tester then run it on the connection.
 I had another drop near by and just swicthed to it.

What kind of test tool would you suggest? Usually we rely on the cabling
guys for that but that entails a delay and I'd be interested in knowing
how to do it myself.

Thanks,
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Nortex Networks
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[asterisk-users] automated dialout detect forward

2007-03-21 Thread Mike Heininger


Hi!

I have an automated dialout via a call file to a mobile.
Can I detect when the call is not answered but forwarded to the mobile
operator voicebox?
I would like to stop the dialout if this is the case.


TIA,
Mike
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[asterisk-users] About Pickup Grandstream

2007-03-21 Thread LKS GMAIL

Greetings to everybody.

 

My question is that its impossible to pick up a call from ZAP, IAX or mISDN
with my Ext Key of my GrandStream.

 

It always give me a Spawn Message on CLI and a 603 error on my LCD
GrandStream.

Exactly from my CLI screen i get this message

 

-- Executing NoOp(SIP/11-096c2ac0, Probando 1 ) in new stack

-- Executing Pickup(SIP/11-096c2ac0, IAX2/panoramix/) in new stack

  == Spawn extension (especial, **13, 2) exited non-zero on
'SIP/11-096c2ac0'

*   Executing NoOp(SIP/11-096c2ac0, causa del colgado: 0) in new
stack

 

Let me explain this one : SIP 11 is the first one that receive the call,
then, 13 is mine one, and IAX2/panoramix the trunk where i receive calls
from.

 

Any idea, please? Sorry about my english.

 

 

Saludos, Lukassky.

 

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[asterisk-users] FWD outgoing problem

2007-03-21 Thread Bogdan GONCIULEA


I have configured iax.conf and extensions.conf as instructed on FWD website
(http://www.freeworlddialup.com/help/?p=knowledgebasec=18a=76) and I can
successfully receive calls and make test calls to 612, 613, etc.
The problem is that I can not make a call to another FWD user. Here is what
asterisk says:

-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1, CALLERID(all)=xx)
in new stack
   -- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, 
IAX2/yy:[EMAIL PROTECTED]/xx|60|r) in new stack
   -- Called yy:[EMAIL PROTECTED]/xx
   -- Call accepted by 192.246.69.186 (format ulaw)
   -- Format for call is ulaw
   -- IAX2/192.246.69.186:4569-1 is busy
   -- Hungup 'IAX2/192.246.69.186:4569-1'
 == Everyone is busy/congested at this time (1:1/0/0)
   -- Executing [EMAIL PROTECTED]:3] Congestion(Zap/1-1, ) in new
stack
 == Spawn extension (default, 393xx, 3) exited non-zero on 'Zap/1-1'
   -- Hungup 'Zap/1-1'

xx - FWD number I want to call
yy - FWD number used by asterisk to register
ppp - password for yy


Thanks,
Bogdan
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Re: [Asterisk-Users] Sipura SPA-841 and firewall

2007-03-21 Thread Wilson Pickett


This old post just saved my nerves! Please remember to post solutions
when you find them as this person did.

SPA-941, swapping in a D-Link switch and DSL modem to replace a
Linksys WAG54-G. This router is notorious for suddently losing the
ability to negotiate bit rates (thanks again, Google).

Upon replacement of the Linksys, everything worked fine except audio
on the Sipura. Turns out you need Symmetric RTP turned on in the phone
as Chris Mason says below.

On 4/29/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:

Just for future reference, I found the answer - I enabled Symmetric RTP: on
the Advanced SIP page.

Chris Mason
www.anguillaguide.com


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Chris Mason (Lists)
 Sent: Thursday, April 28, 2005 6:22 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] Sipura SPA-841 and firewall

 I have an asterisk server and 4 Sipura phones behind a
 Linksys WRT54G router. I have set the DMZ to the Asterisk
 server's IP so that it can be seen from outside. I have a
 Sipura SPA-841 phone outside the router and set to proxy to
 the public IP of the router. The outside phone registers
 fine, dials fine, and I can hear the person speaking from
 inside the router, but I cannot be heard.

 Is there any explanation for this? Surely the DMZ allows all
 traffic to the PBX?
 This is driving me nuts.

 Chris Mason

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Re: [asterisk-users] snom led not working with asterisk 1.4.1

2007-03-21 Thread Giorgio Incantalupo


Hi Steve,
thank you for your help, I set up the call-limit parameter  and SNOM 
light are working good for ringing and busy status. I took a look at 
sip.conf.sample but nothing about unavailable status. Should I set some 
other parameter or there is some trick? Consider that my firmware 
phone is updated to the last available version.


TIA

Giorgio Incantalupo



Steve Murphy wrote:

On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote:
  

Hi,
I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to 
show which devices are busy/not connected. The same phone worked with 
Asterisk 1.2.9.1.
I would appreciate anyone who knows how to setup Asterisk 1.4.1 to 
behave as 1.2.9.1.



Giorgio--

That's a pretty generic question! But that aside, there's been a
substantive change in the configs for SIP phones, that could easily
affect your device state monitoring.

So, suggestion: read the example sip config file in the src/configs dir,
pay close attention to stuff like call-limit, the limitonpeers stuff,
etc, and then make sure you update all your phone entries in sip.conf.
Restart asterisk, or reload sip, and hopefully your lights will work.

In general, EVERYONE, here's some advise: When you 
upgrade from version 1.x to 1.(x+2), always review ALL

your config files against the new config file examples.
Things change! Hopefully, for the better!

murf

  


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Re: [asterisk-users] FWD outgoing problem

2007-03-21 Thread Alex Robar


Can you post the portion of extensions.conf where your Dial command is for
FWD? From the output there it looks like you're trying to dial a FWD number
from a Zap trunk.

Alex

On 3/21/07, Bogdan GONCIULEA [EMAIL PROTECTED] wrote:


I have configured iax.conf and extensions.conf as instructed on FWD
website (http://www.freeworlddialup.com/help/?p=knowledgebasec=18a=76 )
and I can successfully receive calls and make test calls to 612, 613, etc.
The problem is that I can not make a call to another FWD user. Here is
what asterisk says:

-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
CALLERID(all)=xx) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, 
IAX2/yy:[EMAIL PROTECTED]/xx|60|rhttp://IAX2/yy:[EMAIL 
PROTECTED]/xx%7C60%7Cr)
in new stack
-- Called yy:[EMAIL PROTECTED]/xx
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-1 is busy
-- Hungup 'IAX2/192.246.69.186:4569-1'
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing [EMAIL PROTECTED]:3] Congestion(Zap/1-1, ) in new
stack
  == Spawn extension (default, 393xx, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'

xx - FWD number I want to call
yy - FWD number used by asterisk to register
ppp - password for yy


Thanks,
Bogdan

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[EMAIL PROTECTED]
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Re: [asterisk-users] High Pitched Noise

2007-03-21 Thread Rob Schall

Doubtful. I'm using the hardware echo cancel on the card. Any other
reasons the calls would get that noise and/or dropped right after
hearing the noise?

Rob


Eric ManxPower Wieling wrote:
 Could you be having ECFO?  See:

 http://lists.digium.com/pipermail/asterisk-dev/2006-August/022062.html
 http://lists.digium.com/pipermail/asterisk-dev/2006-August/022111.html

 Rob Schall wrote:
 This is a PRI 24 channel line. We have backup pots lines, but they
 aren't in use. The problem we were having was happening on only a single
 channel or 2.

 Rob


 Noah Miller wrote:
 Hi Rob -

 After about having the server running for about an hour, our callers
 occationally hear a high pitched beep that lasts the entire call. In
 some cases, the noise doesn't start until a minute or 2 into the call,
 while others last the entire call. In some of the more serious cases,
 calls are dropped after the noise has occurred as well.

 Another symptom has been really bad static on a specific channel.
 After
 reseating the card to try to fix both this as well as the problem
 above,
 the problem usually goes away, but it seems to come back quicker each
 time. Also, the channel that the static occurs on changes after each
 reseating (after some time).
 What kind of PSTN lines are they?  If they're POTS lines, can you plug
 a regular phone in and test the noise then?  Also have you looked at
 other hardware devices inside the asterisk box?  I've heard disk
 drives (hard, floppy, optical) that make loud enough noises that they
 interfere with analog phone lines.  Do you have another machine to
 test the card in?
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Re: [asterisk-users] Limit call duration

2007-03-21 Thread Suity Zsolt


Robert Lister wrote:

On Wed, Mar 21, 2007 at 12:56:55PM +0100, Suity Zsolt wrote:

Hi everyone,

I'm new to Asterisk, but I like it ;o)
Have a question to you;

How can I limit the incoming call duration?


I think you can say something like:

AbsoluteTimeout (or in 1.2x, Set(TIMEOUT(absolute) = seconds) )

See: http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout


Thank you,
I will try later today, but I think this is what I looking for.
(If I can set it only for external calls)

--
Suich
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Re: Re: [asterisk-users] Tesco Internet Phone

2007-03-21 Thread phil . dawson

I'm also interested in finding a driver for this phone.  I did find a link
to the drivers page of the manufacturer of the phone Yamamoto.  See the
link below.  I've also contacted them about drivers for Linux, asterisk
etc.  I'll report back if I get a reply.

http://www.yamamoto-group.co.uk/index.php?page=download



Phil.







   
 Antony Stone  
 [EMAIL PROTECTED] 
 erisk.open.source  To 
 .it  Asterisk Users Mailing List -   
 Sent by:  Non-Commercial Discussion   
 asterisk-users-bo asterisk-users@lists.digium.com   
 [EMAIL PROTECTED]  cc 
 m.com 
   Subject 
   Re: [asterisk-users] Tesco Internet 
 21/03/2007 11:37  Phone   
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




On Friday 02 March 2007 07:46, Alan Chandler wrote:

 On Thursday 01 March 2007 20:33, bails wrote:
  plug it in a linux box and tell us what it is please,
  generic-usb-audio or what?
 
  Bails
 
  Julian Lyndon-Smith wrote:
   Yeah, that's where firefly comes from, doesn't it.
  
   I've got the base station plugged in, and the handset connected to
   it, but it always says pc unavailable.
  
   My system (xp) sees a usb phone for speakers and microphone,
   but I can't get it to work.

 Did this go any further.  I would be interested in this.

I too would really like to find or help adapt a driver for this.

Here's what I get from my USB-DECT device:

# cat /proc/bus/usb/devices
T:  Bus=01 Lev=02 Prnt=02 Port=00 Cnt=01 Dev#=  6 Spd=12  MxCh= 0
D:  Ver= 1.10 Cls=00(ifc ) Sub=00 Prot=00 MxPS=64 #Cfgs=  1
P:  Vendor=19af ProdID=694d Rev= 0.00
S:  Manufacturer=innoMax Technology Ltd.
S:  Product=Cordless USB Phone
C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=400mA
I:  If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=audio
I:  If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 1 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=83(I) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=02(O) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 3 Alt= 0 #EPs= 2 Cls=03(HID  ) Sub=00 Prot=00 Driver=hid
E:  Ad=81(I) Atr=03(Int.) MxPS=  32 Ivl=1ms
E:  Ad=01(O) Atr=03(Int.) MxPS=  32 Ivl=1ms

# lsusb -v -s 006

Bus 001 Device 006: ID 19af:694d
Device Descriptor:
  bLength18
  bDescriptorType 1
  bcdUSB   1.10
  bDeviceClass0 (Defined at Interface level)
  bDeviceSubClass 0
  bDeviceProtocol 0
  bMaxPacketSize064
  idVendor   0x19af
  idProduct  0x694d
  bcdDevice0.00
  iManufacturer   1 innoMax Technology Ltd.
  iProduct2 Cordless USB Phone
  iSerial 0
  bNumConfigurations  1
  Configuration Descriptor:
bLength 9
bDescriptorType 2
wTotalLength  214
bNumInterfaces  4
bConfigurationValue 1
iConfiguration  0
bmAttributes 0x80
MaxPower  400mA
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber0
  bAlternateSetting   0
  bNumEndpoints   0
  bInterfaceClass 1 Audio
  bInterfaceSubClass  1 Control Device
  bInterfaceProtocol  0
  iInterface  0
  AudioControl Interface Descriptor:
bLength10
bDescriptorType36
bDescriptorSubtype  1 (HEADER)
bcdADC   1.00
wTotalLength   60

[asterisk-users] Metaswitch help needed

2007-03-21 Thread Steve Edwards

I'm attempting to connect to a Metaswitch, inbound only (at this time). 
The Metaswitch is the only connection (at this time).


All I'm getting so far is a bunch of OPTION messages which my Asterisk 
box replies to but I don't get inbound calls.


Here's my sip.conf. As you can see I've been trying a bunch of different 
options without success :(


(206.b.c.d is the address of my Asterisk box. 172.b.c.d is the address of 
the Metaswitch)


[general]
 disallow   = all
allguest= yes
allow   = all
allowguest  = yes
autocreatepeer  = yes
autodomain  = yes
bindaddr= 206.b.c.d
bindport= 5060
callerid= metaswitch 
canreinvite = no
context = test
dtmfmode= rfc2833
host= 172.b.c.d
;   insecure= invite
insecure= very
nat = never
;   nat = yes
port= 5060
qualify = yes
qualifysmoothing= yes
realm   = 206.b.c.d
;   realm   = metaswitch
regcontext  = test
secret  = metaswitch
sipdebug= yes
type= friend
;   type= peer
;   type= user
username= metaswitch

Here's the console SIP debug messages:

-- SIP read from 172.b.c.d:5060: 
OPTIONS sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 
172.b.c.d:5060;rport;branch=z9hG4bK-17eb587208b656d9c2fbd516b5e5401e-172.b.c.d-1
Allow-Events: message-summary
Allow-Events: refer
Allow-Events: dialog
Allow-Events: line-seize
Max-Forwards: 70
Call-ID: [EMAIL PROTECTED]
From: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=172.b.c.d+1+0+22022a3b
CSeq: 445762257 OPTIONS
Organization: 
Supported: 100rel

Content-Length: 0
Contact: sip:[EMAIL PROTECTED]:5060;transport=udp
To: sip:[EMAIL PROTECTED]


--- (15 headers 0 lines) ---
Looking for metaswitch in test (domain 206.b.c.d)
Transmitting (no NAT) to 172.b.c.d:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.b.c.d:5060;rport;branch=z9hG4bK-17eb587208b656d9c2fbd516b5e5401e-172.b.c.d-1;received=172.b.c.d
From: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=172.b.c.d+1+0+22022a3b
To: sip:[EMAIL PROTECTED];tag=as6a59273b
Call-ID: [EMAIL PROTECTED]
CSeq: 445762257 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:206.b.c.d
Accept: application/sdp
Content-Length: 0


---
Destroying call '[EMAIL PROTECTED]'

And this is what I get from sudo ngrep -s 2048 port 5060:

U 172.b.c.d:5060 - 206.b.c.d:5060
  OPTIONS sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 
172.b.c.d:5060;rport;branch=z9hG4bK-815d5107
  ec165bef012bcfebc6e214fd-172.b.c.d-1..Allow-Events: 
message-summary..Allow-Events: refer..Allow-Events: dialog..Allow-Events:
  line-seize..Max-Forwards: 70..Call-ID: [EMAIL PROTECTED]: sip:[EMAIL 
PROTECTED]:5060;transport=udp;tag=172.b.c.d
  +1+0+85ece24c..CSeq: 528990954 OPTIONS..Organization: ..Supported: 100rel..Content-Length: 0..Contact: sip:[EMAIL PROTECTED] .2:5060;transport=udp..To: sip:[EMAIL PROTECTED] 
#

U 206.b.c.d:5060 - 172.b.c.d:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 
172.b.c.d:5060;rport;branch=z9hG4bK-815d5107ec165bef012bcfebc6e214fd-172.b.c.d-1;received=17
  2.16.1.2..From: sip:[EMAIL 
PROTECTED]:5060;transport=udp;tag=172.b.c.d+1+0+85ece24c..To: sip:[EMAIL 
PROTECTED];
  tag=as26804e9e..Call-ID: [EMAIL PROTECTED]: 528990954 OPTIONS..User-Agent: 
Asterisk PBX..Allow: INVITE, ACK, CANCEL, OP
  TIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:206.b.c.d..Accept: application/sdp..Content-Length: 0 
#


Any clues will be appreciated :)

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] snom led not working with asterisk 1.4.1

On Wed, 2007-03-21 at 13:55 +0100, Giorgio Incantalupo wrote:
 Hi Steve,
 thank you for your help, I set up the call-limit parameter  and SNOM 
 light are working good for ringing and busy status. I took a look at 
 sip.conf.sample but nothing about unavailable status. Should I set some 
 other parameter or there is some trick? Consider that my firmware 
 phone is updated to the last available version.
 
 TIA
 
 Giorgio Incantalupo
 
 
Giorgio--

no tricks, sorry!... I've got a snom360 here, and I've been slowly
working my way thru the buttons myself. There's a config file option to
make the Retrieve button work, you provide a name for an extension for
it to use. You then provide
that extension in the context for the phone, that does the
VoiceMailMain() call.

The Record button uses a SIP INFO message to asterisk, that isn't
implemented, so that's not going to work at the moment.

What does unavailable mean, and how do you get that way?

murf

 
 Steve Murphy wrote:
  On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote:

  Hi,
  I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to 
  show which devices are busy/not connected. The same phone worked with 
  Asterisk 1.2.9.1.
  I would appreciate anyone who knows how to setup Asterisk 1.4.1 to 
  behave as 1.2.9.1.
  
 
  Giorgio--
 
  That's a pretty generic question! But that aside, there's been a
  substantive change in the configs for SIP phones, that could easily
  affect your device state monitoring.
 
  So, suggestion: read the example sip config file in the src/configs dir,
  pay close attention to stuff like call-limit, the limitonpeers stuff,
  etc, and then make sure you update all your phone entries in sip.conf.
  Restart asterisk, or reload sip, and hopefully your lights will work.
 
  In general, EVERYONE, here's some advise: When you 
  upgrade from version 1.x to 1.(x+2), always review ALL
  your config files against the new config file examples.
  Things change! Hopefully, for the better!
 
  murf
 

 
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-- 
Steve Murphy
Software Developer
Digium


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[asterisk-users] wct4xxp problem

2007-03-21 Thread Matija Turk


I have one 120 channels isdn pri digium card. I worked fine for the
last 3 months. There was a power outage, and after the server came up
and isdn modem from the telecom, zaptel can't detect connection
(alarms are red). Currently I can't test with another card to be sure,
but is it possible that the power outage burned the card somehow? UPS
was connected, but without controll cable, so after it lost power the
server lost power too (without shutdown).

This happened to me before, but I didn't have the time to analyze and
I just put another card in the server. This is the second time now.

Any ideas on what to do?
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Re: [asterisk-users] snom led not working with asterisk 1.4.1

2007-03-21 Thread Andrew Latham


I think he meant DND, you can program the DND to send Asterisk a call
like *79 or something like that



On 3/21/07, Steve Murphy [EMAIL PROTECTED] wrote:

On Wed, 2007-03-21 at 13:55 +0100, Giorgio Incantalupo wrote:
 Hi Steve,
 thank you for your help, I set up the call-limit parameter  and SNOM
 light are working good for ringing and busy status. I took a look at
 sip.conf.sample but nothing about unavailable status. Should I set some
 other parameter or there is some trick? Consider that my firmware
 phone is updated to the last available version.

 TIA

 Giorgio Incantalupo


Giorgio--

no tricks, sorry!... I've got a snom360 here, and I've been slowly
working my way thru the buttons myself. There's a config file option to
make the Retrieve button work, you provide a name for an extension for
it to use. You then provide
that extension in the context for the phone, that does the
VoiceMailMain() call.

The Record button uses a SIP INFO message to asterisk, that isn't
implemented, so that's not going to work at the moment.

What does unavailable mean, and how do you get that way?

murf


 Steve Murphy wrote:
  On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote:
 
  Hi,
  I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to
  show which devices are busy/not connected. The same phone worked with
  Asterisk 1.2.9.1.
  I would appreciate anyone who knows how to setup Asterisk 1.4.1 to
  behave as 1.2.9.1.
 
 
  Giorgio--
 
  That's a pretty generic question! But that aside, there's been a
  substantive change in the configs for SIP phones, that could easily
  affect your device state monitoring.
 
  So, suggestion: read the example sip config file in the src/configs dir,
  pay close attention to stuff like call-limit, the limitonpeers stuff,
  etc, and then make sure you update all your phone entries in sip.conf.
  Restart asterisk, or reload sip, and hopefully your lights will work.
 
  In general, EVERYONE, here's some advise: When you
  upgrade from version 1.x to 1.(x+2), always review ALL
  your config files against the new config file examples.
  Things change! Hopefully, for the better!
 
  murf
 
 

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--
Steve Murphy
Software Developer
Digium

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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [asterisk-users] FWD outgoing problem

2007-03-21 Thread Wilson Pickett


Dial(Zap/1-1,IAX2/yy:[EMAIL PROTECTED]/xx|60|r)


Looks right to me and the call seems to be accepted by FWD. What
codecs are you using?
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Re: [asterisk-users] snom led not working with asterisk 1.4.1

2007-03-21 Thread Giorgio Incantalupo


Hi Steve,
as you know if you type show hints inside asterisk console you can see 
phone status. When a phone is not connected, Asterisk says it is 
Unavailable. With Asterisk 1.2.9.1 my SNOM leds worked well so I knew 
when a phone was not available but with Asterisk 1.4.1 is not possible 
anymore. This is one of the functions which I'm trying to keep from 
Asterisk 1.2.9.1 to 1.4.1 .


TIA

Giorgio Incantalupo



Steve Murphy wrote:

On Wed, 2007-03-21 at 13:55 +0100, Giorgio Incantalupo wrote:
  

Hi Steve,
thank you for your help, I set up the call-limit parameter  and SNOM 
light are working good for ringing and busy status. I took a look at 
sip.conf.sample but nothing about unavailable status. Should I set some 
other parameter or there is some trick? Consider that my firmware 
phone is updated to the last available version.


TIA

Giorgio Incantalupo




Giorgio--

no tricks, sorry!... I've got a snom360 here, and I've been slowly
working my way thru the buttons myself. There's a config file option to
make the Retrieve button work, you provide a name for an extension for
it to use. You then provide
that extension in the context for the phone, that does the
VoiceMailMain() call.

The Record button uses a SIP INFO message to asterisk, that isn't
implemented, so that's not going to work at the moment.

What does unavailable mean, and how do you get that way?

murf

  

Steve Murphy wrote:


On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote:
  
  

Hi,
I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to 
show which devices are busy/not connected. The same phone worked with 
Asterisk 1.2.9.1.
I would appreciate anyone who knows how to setup Asterisk 1.4.1 to 
behave as 1.2.9.1.



Giorgio--

That's a pretty generic question! But that aside, there's been a
substantive change in the configs for SIP phones, that could easily
affect your device state monitoring.

So, suggestion: read the example sip config file in the src/configs dir,
pay close attention to stuff like call-limit, the limitonpeers stuff,
etc, and then make sure you update all your phone entries in sip.conf.
Restart asterisk, or reload sip, and hopefully your lights will work.

In general, EVERYONE, here's some advise: When you 
upgrade from version 1.x to 1.(x+2), always review ALL

your config files against the new config file examples.
Things change! Hopefully, for the better!

murf

  
  

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[asterisk-users] Asterisk with AudioCodes Mediant 2000

2007-03-21 Thread Kinjal Dixit


hi

I am trying to get Asterisk to work with Mediant 2000.  I have searched 
and found a few articles on the topic, but none of them seem to solve my 
problem.  My knowledge is weaker on the gateway side.  To begin with, I 
think I should get Softphone to dial through the gateway.  I need to 
know how to configure the Mediant 2000 so it can act as a SIP proxy.  
The asterisk side would be easy enough to figure out.


regards
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[asterisk-users] How to get AEL2

2007-03-21 Thread Rizwan Hisham


Hi,
im using asterisk 1.4.2. i need to use the AEL2. its not included in 1.4.2.
how do i get it? is there any patch for this?

--
Regards
Rizwan Hisham
Software Engineer
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[asterisk-users] Asterisk 1.2.17 Released

2007-03-21 Thread Asterisk Development Team

The Asterisk and Zaptel development teams have released Asterisk version
1.2.17.

Along with minor bug fixes, this release incorporates a fix for the SIP
DoS vulnerability recently discovered by INRIA Lorraine
(http://voipsa.org/pipermail/voipsec_voipsa.org/2007-March/002275.html).

All users of Asterisk 1.2 with the SIP channel driver loaded and
connected to an untrusted network are urged to update to this release to
avoid the possibility of experiencing this problem.

Thanks for your support of Asterisk and Zaptel!

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[asterisk-users] Asterisk 1.4.2 Released

2007-03-21 Thread Asterisk Development Team

The Asterisk and Zaptel development teams have released Asterisk 1.4.2.

In addition to minor bug fixes, this release includes:

- improved SLA support, sample configurations and documentation

- fixes for incoming DTMF handling in the IAX2 channel driver

There are also two security-related changes in this version:

- a fix for a SIP channel driver remote DoS vulnerability
(http://bugs.digium.com/view.php?id=9313)

- a fix for a SIP channel driver remote DoS vulnerability discovered by
INRIA Lorraine
(http://voipsa.org/pipermail/voipsec_voipsa.org/2007-March/002275.html)

All users of Asterisk 1.4 with the SIP channel driver loaded and
connected to an untrusted network are urged to update to this release to
avoid the possibility of experiencing these problems.

Thanks for your support of Asterisk and Zaptel!

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[asterisk-users] reducing the number of extensions for every user

2007-03-21 Thread Rizwan Hisham


Hi,
here is my scenario

in my voip system(asterisk based) every user has a primary did and 5
secondary did's i.e. all six did's point to a single channel. every user has
a blacklist feature and a call filter feature. if blacklist feature is
enabled, user has option to include 5 bl numbers. if the user is using all 5
bl numbers then i have to match DNID with every(6) did. im not using
asterisk DB so i cant use LookupBlackList. if i have 15,000 users subscribed
and using our services then the number of extensions just multiplies with
30. I think thats a lot of extensions. so i need to find a way to reduce the
number of extensions.

i mean currently im doing this

;primary did
exten= 1,1,Dial(SIP/riz)

;sec did's
exten= 2,1,Goto(,1,1)
exten= 3,1,Goto(,1,1)
exten= 4,1,Goto(,1,1)
exten= 5,1,Goto(,1,1)
exten= 6,1,Goto(,1,1)

if blacklist is enabled and user has added 5 bl numbers then.
exten= 1,1,Dial(SIP/riz)
exten= 1/20,1,Hangup
exten= 1/21,1,Hangup
exten= 1/22,1,Hangup
exten= 1/23,1,Hangup
exten= 1/24,1,Hangup

same goes for every secondary did. and situation is even worse if the user
enables the call filter feature i.e. same thing has to be done for cf
numbers. im thinking a way to avoid this and reduce extensions and in the
mean time i need your help.

--
Regards
Rizwan Hisham
Software Engineer
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Re: [asterisk-users] How to get AEL2

2007-03-21 Thread Sean Bright


The latest release of Asterisk is 1.4.1, or am I missing something?

On 3/21/07, Rizwan Hisham [EMAIL PROTECTED] wrote:


Hi,
im using asterisk 1.4.2. i need to use the AEL2. its not included in 1.4.2.
how do i get it? is there any patch for this?

--
Regards
Rizwan Hisham
Software Engineer
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--
sean
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[asterisk-users] FWD outgoing problem

2007-03-21 Thread Bogdan Gonciulea

[globals]
FWDNUMBER=yy
FWDPASSWORD=
FWDCIDNAME=some name

[default]
exten = _393.,1,Set(CALLERID(all)=${FWDCIDNAME})
exten =
_393.,n,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,
r)
exten = _393.,n,Congestion


I have also took out the Set(CALLERID...) line and the result was the same.

Indeed I have a configured Zap trunk which I use to place calls. As I said
making calls to 612(time) and 613(echo) - dialing 393612 or 393613 - works
very well. The problem is when trying to call another FWD user (dialing
393xx on my phone) I get the busy signal, asterisk saying
IAX2/192.246.69.186:4569-1 is busy.

I found on the FWD forums that others had the same problem, but I couldn't
find any solution. They also said that in fact the problem appears only when
asterisk (which connects to FWD using IAX) is trying to call a FWD user
which is using SIP. If I try to call a FWD user which is registered from
behind another asterisk system which is connected to FWD through IAX it
should work. I didn't test this yet...

Bogdan

-- 
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 268.18.16/729 - Release Date: 3/21/2007
7:52 AM
 

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Re: [asterisk-users] FWD outgoing problem

2007-03-21 Thread Derek Whitten

Wilson Pickett wrote:
 Dial(Zap/1-1,IAX2/yy:[EMAIL PROTECTED]/xx|60|r)
 
 Looks right to me and the call seems to be accepted by FWD. What
 codecs are you using?
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I have been having issues with FWD lately (basically nothing has worked on FWD 
for a few
days now)





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Re: [asterisk-users] automated dialout detect forward

2007-03-21 Thread Robert Lister

On Wed, Mar 21, 2007 at 01:23:37PM +0100, Mike Heininger wrote:
 Hi!
 
 I have an automated dialout via a call file to a mobile.
 Can I detect when the call is not answered but forwarded to the mobile
 operator voicebox?
 I would like to stop the dialout if this is the case.
 

One simple method would be to dial out and then playback an announcement 
announcing the incoming call, maybe even the number, and ask the user to 
press some key to accept the call. If this key is not pressed within a 
certain timeout, then terminate. This is okay to detect answering machines
etc.

I believe asterisk 1.4. has some better controls over this.

In 1.2, some other techniques are discussed at:

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackgroundDetect


Rob

-- 
Robert Lister   -   London Internet Exchange-  http://www.linx.net/
[EMAIL PROTECTED]   -   tel: +44 (0)20 7645 3510-  RL786-RIPE
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Re: [asterisk-users] Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation

2007-03-21 Thread John Novack




George C. Attopany wrote:


Hi,

I run ASTERISK  1.2  with a  Wildcard TE410P-Xilinx on Redhat Linux 
8.0(Psyche) with Kernel 2.4.18-14 on an i686 with a Channel bank to 
enable me use anlog handsets.


I would like to upgrade to Fedora Core 6 and then run  ASTERISK 1.4.1 
and latest Zaptel etc.



Why not use CentOS 4.4 instead?
The price is right, and it seems there are few problems with CentOS vs 
Fedora.


John Novack

Reading from the mailing list about problems with Fedora and Linux 
Kernel 2.6,  and from the link ( 
http://www.voip-info.org/wiki/index.php?page=Asterisk+OS+Platforms ) I 
am a bit confused about the way forward with my upgrade.


I am not very good with Linux systems, but would appreciate your 
advice to sail through my upgrade successful. Any hint from you would 
be welcome, in particular how to upgrade my platform to Fedora Core 6 
and the necessary Kernel to give me a stable platform.


Looking forward to hearing from you.


Kind regards,

george.

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[asterisk-users] asterisk log analyzer

2007-03-21 Thread Eric Smith

Please point me to a simple analyzer tool that parses asterisk log
files.  I do not want a web based or java application,
Just a script that parses the logs and extracts highest priority
information, ideally something I can put in a pipe (not to smoke).

Thanks

-- 
Eric Smith
Fruitcom Amsterdam
Tel: +31 20 4111 834
Fax: +31 20 4114 619
www.fruitcom.com
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Re: [asterisk-users] How to get AEL2

2007-03-21 Thread Sean Bright


Apparently I was missing something :-)  Just saw the mailing list message
about 1.4.2.

On 3/21/07, Sean Bright [EMAIL PROTECTED] wrote:


The latest release of Asterisk is 1.4.1, or am I missing something?

On 3/21/07, Rizwan Hisham [EMAIL PROTECTED]  wrote:

 Hi,
 im using asterisk 1.4.2. i need to use the AEL2. its not included in
 1.4.2. how do i get it? is there any patch for this?

 --
 Regards
 Rizwan Hisham
 Software Engineer
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--
sean





--
sean
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[asterisk-users] ses ActiveDirectory and also Ldap and Kerberos.

2007-03-21 Thread Carlos Jerónimo


Hi i'm student and my final project is related to Voip. I have
Asterisk almost fully configured. The next step is to accept login of
users, that data is in Universitys database which uses ActiveDirectory
and also Ldap and Kerberos.
It's possible? I don't want authentications in sip.conf, but in other
remote database.
The problem is i don't have ideas how to start with.

I would appreciate some ideas

be
--
Carlos Jerónimo
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RE: Re: [asterisk-users] Tesco Internet Phone

2007-03-21 Thread asterisk

I use this driver for the SJ phone with the USB tesco internet phone:

http://www.sjphone.org/usbphone/SJphoneDriverWebPoint.exe

Fadge


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 21 March 2007 13:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re: [asterisk-users] Tesco Internet Phone

I'm also interested in finding a driver for this phone.  I did find a link
to the drivers page of the manufacturer of the phone Yamamoto.  See the
link below.  I've also contacted them about drivers for Linux, asterisk
etc.  I'll report back if I get a reply.

http://www.yamamoto-group.co.uk/index.php?page=download



Phil.







   
 Antony Stone  
 [EMAIL PROTECTED] 
 erisk.open.source  To 
 .it  Asterisk Users Mailing List -   
 Sent by:  Non-Commercial Discussion   
 asterisk-users-bo asterisk-users@lists.digium.com   
 [EMAIL PROTECTED]  cc 
 m.com 
   Subject 
   Re: [asterisk-users] Tesco Internet 
 21/03/2007 11:37  Phone   
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




On Friday 02 March 2007 07:46, Alan Chandler wrote:

 On Thursday 01 March 2007 20:33, bails wrote:
  plug it in a linux box and tell us what it is please,
  generic-usb-audio or what?
 
  Bails
 
  Julian Lyndon-Smith wrote:
   Yeah, that's where firefly comes from, doesn't it.
  
   I've got the base station plugged in, and the handset connected to
   it, but it always says pc unavailable.
  
   My system (xp) sees a usb phone for speakers and microphone,
   but I can't get it to work.

 Did this go any further.  I would be interested in this.

I too would really like to find or help adapt a driver for this.

Here's what I get from my USB-DECT device:

# cat /proc/bus/usb/devices
T:  Bus=01 Lev=02 Prnt=02 Port=00 Cnt=01 Dev#=  6 Spd=12  MxCh= 0
D:  Ver= 1.10 Cls=00(ifc ) Sub=00 Prot=00 MxPS=64 #Cfgs=  1
P:  Vendor=19af ProdID=694d Rev= 0.00
S:  Manufacturer=innoMax Technology Ltd.
S:  Product=Cordless USB Phone
C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=400mA
I:  If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=audio
I:  If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 1 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=83(I) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
I:  If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
E:  Ad=02(O) Atr=01(Isoc) MxPS=  64 Ivl=1ms
I:  If#= 3 Alt= 0 #EPs= 2 Cls=03(HID  ) Sub=00 Prot=00 Driver=hid
E:  Ad=81(I) Atr=03(Int.) MxPS=  32 Ivl=1ms
E:  Ad=01(O) Atr=03(Int.) MxPS=  32 Ivl=1ms

# lsusb -v -s 006

Bus 001 Device 006: ID 19af:694d
Device Descriptor:
  bLength18
  bDescriptorType 1
  bcdUSB   1.10
  bDeviceClass0 (Defined at Interface level)
  bDeviceSubClass 0
  bDeviceProtocol 0
  bMaxPacketSize064
  idVendor   0x19af
  idProduct  0x694d
  bcdDevice0.00
  iManufacturer   1 innoMax Technology Ltd.
  iProduct2 Cordless USB Phone
  iSerial 0
  bNumConfigurations  1
  Configuration Descriptor:
bLength 9
bDescriptorType 2
wTotalLength  214
bNumInterfaces  4
bConfigurationValue 1
iConfiguration  0
bmAttributes 0x80
MaxPower  400mA
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber0
  bAlternateSetting   0
  bNumEndpoints

[asterisk-users] res_musiconhold.c:1243 load_module: No music on hold classes configured

2007-03-21 Thread Thomas Winter

Hi,

I am using relatime for musiconhold.conf.

After starting Asterisk I have to do an reload, otherwise no MoH is avaiable.

Bug or do I have to change loading of modules in modules.con?

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Re: [asterisk-users] How to get AEL2

2007-03-21 Thread Rizwan Hisham


its on the ftp link here ftp://ftp.digium.com/pub/asterisk , it was put on
yesterday

On 3/21/07, Sean Bright [EMAIL PROTECTED] wrote:


The latest release of Asterisk is 1.4.1, or am I missing something?

On 3/21/07, Rizwan Hisham [EMAIL PROTECTED]  wrote:

 Hi,
 im using asterisk 1.4.2. i need to use the AEL2. its not included in
 1.4.2. how do i get it? is there any patch for this?

 --
 Regards
 Rizwan Hisham
 Software Engineer
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--
sean
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--
Regards
Rizwan Hisham
Software Engineer
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[asterisk-users] PickUp a call with feature pickup (*8) from a IAX2 channel

2007-03-21 Thread Alvaro Parres


Hi list, i'm trying to do that iax channels can acces the pickup
feature(normaly *8 dialing).

But always the iax channel when dial *8, search for the extensión *8 on its
context.

I know i can program the *8 extension with the pickup applicatión. But its
doesn't works for me, becouse i need to pickup some calls comming from IVR's
o Queues.
And there de exten is no the same as the channel, etc.

Any idea or help ?

Thaks.
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Re: [asterisk-users] FWD outgoing problem

2007-03-21 Thread Robert Lister

On Wed, Mar 21, 2007 at 04:40:02PM +0200, Bogdan Gonciulea wrote:
 [globals]
 FWDNUMBER=yy
 FWDPASSWORD=
 FWDCIDNAME=some name
 
 [default]
 exten = _393.,1,Set(CALLERID(all)=${FWDCIDNAME})
 exten = _393.,n,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
 exten = _393.,n,Congestion
 
 I have also took out the Set(CALLERID...) line and the result was the same.

That doesn't look entirely right to me, maybe it should be:

Set(CALLERID(name)=${FWDCIDNAME})
Set(CALLERID(num)=${FWDNUMBER})

Set(CALLERID(all)= is for setting the entire caller ID header, so it
should look something like this if you use it:

Set(CALLERID(all)=Joe User 1234)

I think the things after DIAL(IAX2/..) should match what you have configured 
in iax.conf for iax peer:

iax.conf (from some example I found):

[FWDIAXPeer]
type=peer
disallow = all
allow=ulaw ; FWD only do ulaw
host=iax2.fwdnet.net
qualify=300 ; optional of course
secret=secret
context=from-fwd
username=321321

Then: Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:3},45)

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Re: [asterisk-users] How to get AEL2

2007-03-21 Thread John Novack




Sean Bright wrote:

The latest release of Asterisk is 1.4.1, or am I missing something?


Sure are.
1.4.2 release was just posted

JN

On 3/21/07, *Rizwan Hisham* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hi,
im using asterisk 1.4.2. i need to use the AEL2. its not included
in 1.4.2. how do i get it? is there any patch for this?

-- 
Regards

Rizwan Hisham
Software Engineer
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--
sean


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Re: [asterisk-users] wct4xxp problem

2007-03-21 Thread Tim Panton



On 21 Mar 2007, at 13:47, Matija Turk wrote:


I have one 120 channels isdn pri digium card. I worked fine for the
last 3 months. There was a power outage, and after the server came up
and isdn modem from the telecom, zaptel can't detect connection
(alarms are red). Currently I can't test with another card to be sure,
but is it possible that the power outage burned the card somehow? UPS
was connected, but without controll cable, so after it lost power the
server lost power too (without shutdown).

This happened to me before, but I didn't have the time to analyze and
I just put another card in the server. This is the second time now.

Any ideas on what to do?


Ring the Telco and ask them what they see.

I once spent a week struggling with this sort of symptom to
find in the end that the ops guys had got fed up with my
line being in 'alarm' on their console and disabled it at their end.

One phone call later it was re-enabled !

Tim.



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Re: [asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?

2007-03-21 Thread Stephen Bosch

Barzilai Spinak wrote:
 An equally unrealistic expectation would be to require that people write
 RELEVANT and specific Subjects.
 If your question relates to 1.4, put 1.4 somewhere in the Subject, or
 if it relates to unreleased trunk, specify it.
 So you can quickly filter out/in whatever your interests are.
 But as I said... it's an unrealistic expectation.

Much like asking people to start a new thread for a new subject :|

-Stephen-
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Re: [asterisk-users] Refund from SellVoip?

2007-03-21 Thread Stephen Bosch

Brad Templeton wrote:
 On Fri, Mar 16, 2007 at 04:16:21PM -0700, Tom Lynn wrote:
 At this point, I'm simply contacting the State of Washington Attorney
 General's office.  They're ignoring my e-mails and I'm done monkeying
 around.

 
 It makes no sense.   The put together a good system on the tech end,
 Asterisk based, decent call quality and faster call completions than
 any of the other folks I have been trying, at good prices.  And
 then dropped it all on the floor, not responding to calls, emails or
 tickets often for weeks and months, if at all.   Their interface needed
 work but that I can tolerate.   Not being able to reach somebody for
 an urgent problem makes no sense.  
 
 Does anybody know Jed Stafford?  As far as I can tell this ended up
 being a one-man or two-man operation.  It's just sad.

Just like 90% of the VoIP outfits out there.

Lately, it seems it's either our way or the highway outfits like
Vonage, or a couple of dudes in their gonch in a basement. There's very
little in between :(

-Stephen-
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Re: [asterisk-users] High Pitched Noise

2007-03-21 Thread Stephen Bosch

Rob Schall wrote:
 Doubtful. I'm using the hardware echo cancel on the card. Any other
 reasons the calls would get that noise and/or dropped right after
 hearing the noise?

Does it matter whether the echo canceller is hardware or software for
ECFO to happen? It seems to me that if you crank the gain enough you can
make any echo canceller wig out.

-Stephen-
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[asterisk-users] Dlink i2eye

2007-03-21 Thread Bruce Reeves


I have seen this product before and wondered, has anyone connected this to
Asterisk?

http://www.i-2-eye.com/index.html

As far as that goes has anyone seen a set top box video phones that work
with Asterisk?

--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] snom led not working with asterisk 1.4.1

On Wed, 2007-03-21 at 15:00 +0100, Giorgio Incantalupo wrote:
 Hi Steve,
 as you know if you type show hints inside asterisk console you can see 
 phone status. When a phone is not connected, Asterisk says it is 
 Unavailable. With Asterisk 1.2.9.1 my SNOM leds worked well so I knew 
 when a phone was not available but with Asterisk 1.4.1 is not possible 
 anymore. This is one of the functions which I'm trying to keep from 
 Asterisk 1.2.9.1 to 1.4.1 .
 

Pardon my ignorance! I am new in this area. I have not used my SNOM 360
with anything but 1.4. When the monitored extension is busy, the LED is
on; when the extension is ringing, the LED flashes. What does it do for
you in 1.2, when the line is unavailable?

murf

 TIA
 
 Giorgio Incantalupo
 
 
 
 Steve Murphy wrote:
  On Wed, 2007-03-21 at 13:55 +0100, Giorgio Incantalupo wrote:

  Hi Steve,
  thank you for your help, I set up the call-limit parameter  and SNOM 
  light are working good for ringing and busy status. I took a look at 
  sip.conf.sample but nothing about unavailable status. Should I set some 
  other parameter or there is some trick? Consider that my firmware 
  phone is updated to the last available version.
 
  TIA
 
  Giorgio Incantalupo
 
 
  
  Giorgio--
 
  no tricks, sorry!... I've got a snom360 here, and I've been slowly
  working my way thru the buttons myself. There's a config file option to
  make the Retrieve button work, you provide a name for an extension for
  it to use. You then provide
  that extension in the context for the phone, that does the
  VoiceMailMain() call.
 
  The Record button uses a SIP INFO message to asterisk, that isn't
  implemented, so that's not going to work at the moment.
 
  What does unavailable mean, and how do you get that way?
 
  murf
 

  Steve Murphy wrote:
  
  On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote:


  Hi,
  I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to 
  show which devices are busy/not connected. The same phone worked with 
  Asterisk 1.2.9.1.
  I would appreciate anyone who knows how to setup Asterisk 1.4.1 to 
  behave as 1.2.9.1.
  
  
  Giorgio--
 
  That's a pretty generic question! But that aside, there's been a
  substantive change in the configs for SIP phones, that could easily
  affect your device state monitoring.
 
  So, suggestion: read the example sip config file in the src/configs dir,
  pay close attention to stuff like call-limit, the limitonpeers stuff,
  etc, and then make sure you update all your phone entries in sip.conf.
  Restart asterisk, or reload sip, and hopefully your lights will work.
 
  In general, EVERYONE, here's some advise: When you 
  upgrade from version 1.x to 1.(x+2), always review ALL
  your config files against the new config file examples.
  Things change! Hopefully, for the better!
 
  murf
 


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-- 
Steve Murphy
Software Developer
Digium

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[asterisk-users] Cisco 7970 with skinny on * 1.4.1

2007-03-21 Thread Richard Klingler


Evnin' (o;


As chan_sccp is pretty much dead, doesn't compile on FBSD anyway
and isn't supported on * 1.4.x I tried going with chan_skinny...

The Cisco 7970 registers and is being acknowledged by * but that's it...

I see no lines on the 7970 display configured and it is not reachable
or it can't make any outboudn calls...

The docs are pretty non-existent for skinny and the sample configuration
are of no help...


Has any1 got their 7970 to work with * 1.4.x ?


cheers
rick



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[asterisk-users] Asterisk 1.4.2 Requires Zaptel from 1.4 svn branch for zap_chan?

2007-03-21 Thread Stuart Sheldon

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Looks like there is a requirement for a later Zaptel the the 1.4.0
release for the zap_chan driver to build...

Stu

- --
Randomly Generated Fortune Tag:
They spell it da Vinci and pronounce it da Vinchy.  Foreigners
always spell better than they pronounce.
-- Mark Twain
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (GNU/Linux)

iD8DBQFGAWVeK69Y+xPZrWYRAvZHAJ41WUVjxtEyGFlyP2kxooiiYGnVAwCfRTEh
WcpvdqJeWnY/R0moszmH98o=
=HJuz
-END PGP SIGNATURE-
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[asterisk-users] How to resolve CallerID from AudioCodes FXO

2007-03-21 Thread hind habaoui


hi angel.
it is about the CallerId, i have the same problem, did you resolve it???
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Re: [asterisk-users] How to get AEL2

2007-03-21 Thread Sean Bright


Are you sure its not included?  I ran 'make menuselect' and under PBX
Modules its the first thing listed for me.

On 3/21/07, Rizwan Hisham [EMAIL PROTECTED] wrote:


its on the ftp link here ftp://ftp.digium.com/pub/asterisk , it was put on
yesterday

On 3/21/07, Sean Bright [EMAIL PROTECTED] wrote:

 The latest release of Asterisk is 1.4.1, or am I missing something?

 On 3/21/07, Rizwan Hisham  [EMAIL PROTECTED]  wrote:

  Hi,
  im using asterisk 1.4.2. i need to use the AEL2. its not included in
  1.4.2. how do i get it? is there any patch for this?
 
  --
  Regards
  Rizwan Hisham
  Software Engineer
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 --
 sean
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--
Regards
Rizwan Hisham
Software Engineer

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--
sean
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RE: [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2channel

2007-03-21 Thread LKS GMAIL

Try to set the callgroup and pickupgroup up in the IAX conf.

 

Saludos, Lukassky.

  _  

De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Alvaro Parres
Enviado el: miércoles, 21 de marzo de 2007 16:55
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] PickUp a call with feature pickup (*8) from a
IAX2channel

 

Hi list, i'm trying to do that iax channels can acces the pickup
feature(normaly *8 dialing).

But always the iax channel when dial *8, search for the extensión *8 on its
context. 

I know i can program the *8 extension with the pickup applicatión. But its
doesn't works for me, becouse i need to pickup some calls comming from IVR's
o Queues. 
And there de exten is no the same as the channel, etc.

Any idea or help ?

Thaks.

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Re: [asterisk-users] Tesco Internet Phone

On Wednesday 21 March 2007 11:57, bails wrote:

 Antony Stone wrote:
  On Friday 02 March 2007 07:46, Alan Chandler wrote:
  On Thursday 01 March 2007 20:33, bails wrote:
  plug it in a linux box and tell us what it is please,
  generic-usb-audio or what?
 
  Bails
 
  Julian Lyndon-Smith wrote:
  Yeah, that's where firefly comes from, doesn't it.
 
  I've got the base station plugged in, and the handset connected to
  it, but it always says pc unavailable.
 
  My system (xp) sees a usb phone for speakers and microphone,
  but I can't get it to work.
 
  Did this go any further.  I would be interested in this.
 
  I too would really like to find or help adapt a driver for this.
 
  Here's what I get from my USB-DECT device:
 
  # cat /proc/bus/usb/devices
  T:  Bus=01 Lev=02 Prnt=02 Port=00 Cnt=01 Dev#=  6 Spd=12  MxCh= 0
  D:  Ver= 1.10 Cls=00(ifc ) Sub=00 Prot=00 MxPS=64 #Cfgs=  1
  P:  Vendor=19af ProdID=694d Rev= 0.00
  S:  Manufacturer=innoMax Technology Ltd.
  S:  Product=Cordless USB Phone
  C:* #Ifs= 4 Cfg#= 1 Atr=80 MxPwr=400mA
  I:  If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=audio
  I:  If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
  I:  If#= 1 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
  E:  Ad=83(I) Atr=01(Isoc) MxPS=  64 Ivl=1ms
  I:  If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=audio
  I:  If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=audio
  E:  Ad=02(O) Atr=01(Isoc) MxPS=  64 Ivl=1ms
  I:  If#= 3 Alt= 0 #EPs= 2 Cls=03(HID  ) Sub=00 Prot=00 Driver=hid
  E:  Ad=81(I) Atr=03(Int.) MxPS=  32 Ivl=1ms
  E:  Ad=01(O) Atr=03(Int.) MxPS=  32 Ivl=1ms
 
  # lsusb -v -s 006

snipped for brevity

  Let me know if I can help with any other info.
 
 
  Antony.

 Whats the output of dmesg when you plug it in?

hub.c: new USB device 00:07.2-1.1, assigned address 7
usbaudio: device 7 audiocontrol interface 0 has 1 input and 1 output 
AudioStreaming interfaces
usbaudio: device 7 interface 1 altsetting 1 channels 1 framesize 2 configured
usbaudio: valid input sample rate 8000
usbaudio: device 7 interface 1 altsetting 1: format 0x0010 sratelo 8000 
sratehi 8000 attributes 0x00
usbaudio: device 7 interface 2 altsetting 0 does not have an endpoint
usbaudio: device 7 interface 2 altsetting 1 channels 1 framesize 2 configured
usbaudio: valid output sample rate 8000
usbaudio: device 7 interface 2 altsetting 1: format 0x0010 sratelo 8000 
sratehi 8000 attributes 0x00
usbaudio: registered dsp 14,19
usbaudio: constructing mixer for Terminal 4 type 0x0301
usbaudio: warning: found 1 of 0 logical channels.
usbaudio: assuming the channel found is the master channel (got a Philips 
camera?). Should be fine.
usbaudio: registered mixer 14,16
usbaudio: constructing mixer for Terminal 2 type 0x0101
usbaudio: unit 0 not found!
usbaudio: no mixer controls found for Terminal 2
usb_audio_parsecontrol: usb_audio_state at c11f93e0
usb_control/bulk_msg: timeout
: USB HID v1.01 Device [innoMax Technology Ltd. Cordless USB Phone] on 
usb1:7.3

-- 
It wouldn't be a good idea to talk about him behind his back in front of 
him.

 - murble

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   please don't CC me.
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Re: [asterisk-users] Tesco Internet Phone

On Wednesday 21 March 2007 15:11, asterisk wrote:

 I use this driver for the SJ phone with the USB tesco internet phone:

 http://www.sjphone.org/usbphone/SJphoneDriverWebPoint.exe

Yes, but that's a corded phone which plugs into the USB socket.

# cat /proc/bus/usb/devices
P:  Vendor=19af ProdID=694d Rev= 0.00
S:  Manufacturer=innoMax Technology Ltd.
S:  Product=Cordless USB Phone

is a DECT phone where the base station plugs into the USB socket.

http://buy.tescointernetphone.com/details.asp?idProduct=669

Antony.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 [EMAIL PROTECTED]
 Sent: 21 March 2007 13:12
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: Re: [asterisk-users] Tesco Internet Phone

 I'm also interested in finding a driver for this phone.  I did find a link
 to the drivers page of the manufacturer of the phone Yamamoto.  See the
 link below.  I've also contacted them about drivers for Linux, asterisk
 etc.  I'll report back if I get a reply.

 http://www.yamamoto-group.co.uk/index.php?page=download



 Phil.

-- 
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 - Terry Pratchett

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Re: [asterisk-users] How to get AEL2

On Wed, 2007-03-21 at 19:09 +0500, Rizwan Hisham wrote:
 Hi,
 im using asterisk 1.4.2. i need to use the AEL2. its not included in
 1.4.2. how do i get it? is there any patch for this?
 

Rizwan--

Simple. AEL in 1.4 **is** AEL2. The new implementation replaced the old
in that release.

murf


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[asterisk-users] install and setup app_mp4 application

2007-03-21 Thread richard Coco


Hi all,

according to
http://sip.fontventa.com/content/view/15/44/ i have
compiled the mpeg4ip libries without problem. After
copying the app_mp4.c file into de Asterisk apps
directory and changing the Makefile like.

[...]
app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $
${CYGSOLIB} -lodbc

app_mp4.so : app_mp4.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $
${CYGSOLIB} -lmp4 -lmp4v2

ifeq (SunOS,$(shell uname))
app_chanspy.so: app_chanspy.o
$(CC) $(SOLINK) -o $@ $ -lrt
endif
[...]

i get following error.
Mar 21 19:08:22 WARNING[26686]: pbx.c:1720
pbx_extension_helper: No application 'mp4save' for
extension...

it seems that after the recompilation of asterisk no
app_mp4.o/app_mp4.so is created in ../asterisk/apps/.

asterisk# ls apps/app_mp*
apps/app_mp3.c  apps/app_mp3.o  apps/app_mp3.so 
apps/app_mp4.c

has anyone an idea...
thx in advabce...
  


 

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RE: [asterisk-users] Dlink i2eye

2007-03-21 Thread Dean Collins

I played with one a long long time ago when they first came out, it's
relied on a third party server (eg in dlinks data center) so wont work
with asterisk but that was a long time ago so may have changed.

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Wednesday, 21 March 2007 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dlink i2eye

 

I have seen this product before and wondered, has anyone connected this
to Asterisk?

http://www.i-2-eye.com/index.html

As far as that goes has anyone seen a set top box video phones that work
with Asterisk? 

-- 
Bruce Reeves
Nortex Networks 

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Re: [asterisk-users] Limit call duration

2007-03-21 Thread Yuan LIU

From: Suity Zsolt [EMAIL PROTECTED]
Date: Wed, 21 Mar 2007 14:09:20 +0100

Robert Lister wrote:

On Wed, Mar 21, 2007 at 12:56:55PM +0100, Suity Zsolt wrote:

Hi everyone,

I'm new to Asterisk, but I like it ;o)
Have a question to you;

How can I limit the incoming call duration?

You could use L() flag in when dialing the physical end point.

Yuan Liu

I think you can say something like:

AbsoluteTimeout (or in 1.2x, Set(TIMEOUT(absolute) = seconds) )

See: http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout

Thank you,
I will try later today, but I think this is what I looking for.
(If I can set it only for external calls)

--
Suich

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Re: [asterisk-users] automated dialout detect forward

2007-03-21 Thread Lenz



If forwarding happens after a number of rings, you could simply cut off  
the call before the required number of rings happen. Otherwise, I' don't  
think there is a simple way to detect being answered by the voicemail  
versus the intended recipient.

l.


On Wed, 21 Mar 2007 13:23:37 +0100, Mike Heininger [EMAIL PROTECTED]  
wrote:



Hi!

I have an automated dialout via a call file to a mobile.
Can I detect when the call is not answered but forwarded to the mobile
operator voicebox?
I would like to stop the dialout if this is the case.


TIA,
Mike


--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
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Re: [asterisk-users] Metaswitch help needed

On Wed, 2007-03-21 at 06:15 -0700, Steve Edwards wrote:
 I'm attempting to connect to a Metaswitch, inbound only (at this time). 
 The Metaswitch is the only connection (at this time).
 
 All I'm getting so far is a bunch of OPTION messages which my Asterisk 
 box replies to but I don't get inbound calls.
 
 Here's my sip.conf. As you can see I've been trying a bunch of different 
 options without success :(

Good news! I'm not the only guy to try to link his Asterisk box to a
Metaswitch!!

There is a wiki page on voip-info that might be helpful to you--

http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+Metaswitch

If your scenario isn't included, please add it to the wiki when you
figure it out!

murf

-- 
Steve Murphy
Software Developer
Digium

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[asterisk-users] Re: Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation

2007-03-21 Thread Benny Amorsen

 dc == dave cantera [EMAIL PROTECTED] writes:

dc I am a bit confused about the way forward with my upgrade. I am
dc not very good with Linux systems, but would appreciate your advice
dc to sail through my upgrade successful.

Avoiding html in email would be a good start.


/Benny


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[asterisk-users] Asterisk 1.4.2 chan_zap

2007-03-21 Thread Jeremiah Millay


Trying to use:
Asterisk 1.4.2
Zaptel 1.4.0

chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0. 
The changelog has this entry:


* channels/chan_zap.c, configure, configure.ac: If we receive
 ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
 tzafrir) Also, update the configure script to make sure that we
 don't try to build chan_zap if the installed version of zaptel
 does not include ZT_EVENT_REMOVED.


Also a link to bugs.digium.com: http://bugs.digium.com/view.php?id=7256

When I run ./configure for asterisk 1.4.2 part of my output includes the 
following:


checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... yes
checking for ZT_EVENT_REMOVED in zaptel/zaptel.h... no
checking for ZT_TCOP_ALLOCATE in zaptel/zaptel.h... no


Am I to understand that to be able to build chan_zap with this new 
version of asterisk I also need to have a new version of zaptel (which 
has yet to be released) that includes ZT_EVENT_REMOVED in zaptel.h? I'm 
able to compile chan_zap under asterisk 1.4.1 against zaptel 1.4.0 just 
fine.


Jeremiah

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[asterisk-users] Voicemail mailbox number passed in connection?

2007-03-21 Thread Lutgring, Sam

Does anyone know how to configure a SIP phone to pass the mailbox number
to the voicemail service when dialing?  I would like to press the
message waiting lamp and be prompted for my password instead of mailbox
number.  Can this be passed in the set-up call or based on caller-id?
 
Thanks
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Re: [asterisk-users] Voicemail mailbox number passed in connection?

2007-03-21 Thread Jay Moore

I do it by calling my own extension.  If it's me calling me, it passes 
me direct to VoicemailMain.  If it's someone else calling me, it rings 
my phone as normal:


exten = 202,1,GotoIf($[${CALLERIDNUM} = 202] ? 5 : 2)
exten = 202,2,Dial(SIP/jay,10,tT)
exten = 202,3,VoiceMail([EMAIL PROTECTED]|u)
exten = 202,4,HangUp()
exten = 202,5,VoicemailMain([EMAIL PROTECTED])

HTH,
Jay

Lutgring, Sam wrote:

Does anyone know how to configure a SIP phone to pass the mailbox number
to the voicemail service when dialing?  I would like to press the
message waiting lamp and be prompted for my password instead of mailbox
number.  Can this be passed in the set-up call or based on caller-id?
 
Thanks






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Re: [asterisk-users] Voicemail mailbox number passed in connection?

2007-03-21 Thread Time Bandit


Does anyone know how to configure a SIP phone to pass the mailbox number to
the voicemail service when dialing?  I would like to press the message
waiting lamp and be prompted for my password instead of mailbox number.
Can this be passed in the set-up call or based on caller-id?


based on callerID with something like this :

exten = *97,1,Answer
exten = *97,n,Wait(1)
exten = *97,n,VoicemailMain([EMAIL PROTECTED])
exten = *97,n,Hangup()

then just configure your phone to point to *97 (or whatever you choose
as this extension)

hth
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Re: [asterisk-users] Voicemail mailbox number passed in connection?

On Wednesday 21 March 2007 19:54, Lutgring, Sam wrote:

 Does anyone know how to configure a SIP phone to pass the mailbox number
 to the voicemail service when dialing?  I would like to press the
 message waiting lamp and be prompted for my password instead of mailbox
 number.  Can this be passed in the set-up call or based on caller-id?

Caller ID is a simple way to do it.

Make the mailbox number the same as your phone number, then select the mailbox 
based on Caller ID.

It's in some ways more secure, too - it means only you (or at least, only your 
phone) can log in to your mailbox, instead of someone else trying from their 
phone by knowing your mailbox number and guessing your password.


Antony.

-- 
There are only 10 types of people in the world:
those who understand binary notation,
and those who don't.

 Please reply to the list;
   please don't CC me.
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RE: [asterisk-users] Looking for a terminations provider (carrier grade)

2007-03-21 Thread Mattt


| -Original Message-
| From: [EMAIL PROTECTED] 
| [mailto:[EMAIL PROTECTED] On Behalf Of sip
| Sent: Thursday, 22 March 2007 7:38 AM
| To: asterisk-biz@lists.digium.com
| Cc: asterisk-users@lists.digium.com
| Subject: [asterisk-users] Looking for a terminations provider 
| (carrier grade)

| I've posted this on both users and biz as I expect to hear more 
| user testimonials from the users list and more biz proposals 
| from the biz list.

  And not because you have so little regard for the rules / guidelines that
go with posting in these forums? This is not the termination providers'
users list - it's the Asterisk users one ;-)

| If you want to offer your own services, 
| please do it in a private email so as not to clutter the lists. 

  Too late :-/

Cheers,
 Mattt.
 
  - ROMATel - VoIP - http://romatel.net
  - SpotSafe - WiFi Hotspot solution - http://spotsafe.net
 
There are only 10 kinds of people.
Those who understand binary, and those who don't.

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[asterisk-users] G729 'disappears' randomly

2007-03-21 Thread Rajeev Natarajan


All,

I have around 10 opteron 165 servers all running Fedora Core 5 and Asterisk
1.2.x (mostly Asterisk 1.2.16) with 15-25 channels of g729 each. They
register without any problem but I had to use the codec_g729.so
corresponding to the i386 version in all of them (asterisk would not start
if i tried the opteron specific one).

The problem: In one of the servers, we seem to lose the registration after a
restart - 'show g729' simply does not work! A restart (or two!) of the
server after that and things seem fine and show g729 shows the correct
number of channels registered. This happens fairly randomly and have no clue
why this happens only on this one machine. What we've tried:
1. permissions on codec_g729a.so - they seem fine
2. overwriting the .so file and restarting asterisk - doesn't work
3. restarting asterisk a few times - doesn't work
4. permissions on .lic file - they seem fine

but none of the above seem to work. The only resolution seems to be to keep
our fingers crossed while we restart the server!

Ideas / thoughts more than welcome!

thanks
rajeev
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Re: [asterisk-users] Re: Fedora + Linux Kernel 2.6 for Zaptel/Asterisk Installation

2007-03-21 Thread Derek Whitten

Benny Amorsen wrote:

 Avoiding html in email would be a good start.


ROFLMAO!



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Description: OpenPGP digital signature
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RE: [asterisk-users] Looking for a terminations provider (carrier grade)

2007-03-21 Thread sip

While it may not be strictly Asterisk, we use Asterisk to connect to
terminations providers. I'm not entirely sure how it's off-topic to ask if
anyone has decent advice on which providers might be easy to integrate with. I
could have sent a separately-worded message to both biz and users lists, but
invariably then someone would also complain that I shouldn't have sent a
similar message to both lists when only one would have done. 

You can't please everyone. But thank you for your ever-so-constructive input. 

N. 



On Thu, 22 Mar 2007 08:22:17 +1000, Mattt wrote
 | -Original Message-
 | From: [EMAIL PROTECTED] 
 | [mailto:[EMAIL PROTECTED] On Behalf Of sip
 | Sent: Thursday, 22 March 2007 7:38 AM
 | To: asterisk-biz@lists.digium.com
 | Cc: asterisk-users@lists.digium.com
 | Subject: [asterisk-users] Looking for a terminations provider 
 | (carrier grade)
 
 | I've posted this on both users and biz as I expect to hear more 
 | user testimonials from the users list and more biz proposals 
 | from the biz list.
 
   And not because you have so little regard for the rules / 
 guidelines that go with posting in these forums? This is not the 
 termination providers' users list - it's the Asterisk users one 
 ;-)
 
 | If you want to offer your own services, 
 | please do it in a private email so as not to clutter the lists.
 
   Too late :-/
 
 Cheers,
  Mattt.
 
   - ROMATel - VoIP - http://romatel.net
   - SpotSafe - WiFi Hotspot solution - http://spotsafe.net
 
 There are only 10 kinds of people.
 Those who understand binary, and those who don't.
 
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[asterisk-users] Looking for a terminations provider (carrier grade)

2007-03-21 Thread sip

We've been using RNK Telecom for terminations for our SIP service, but their
billing is questionable, they've been in breach of contract multiple times,
and when I brought it to their attention, Russ Man, our 'friendly' account
manager told me, and I quote If you are having that much of a problem..please
find another carrier.

With customer service like that, I've no doubt we could do better, but, while
I have a few ideas in mind, I'd like to hear overall experience with some of
the carrier-grade providers. 

We're looking for several things: 

-Quality connections globally (we provide global service to our customers)
-Ease of integration
-Good rates
-Excellent customer service (although, compared to RNK, if they have a trained
monkey working the phones, they can probably impress us)
-Flexible/no contracts


Any and all experience you have would be good to hear. I've posted this on
both users and biz as I expect to hear more user testimonials from the users
list and more biz proposals from the biz list. If you want to offer your own
services, please do it in a private email so as not to clutter the lists. 

N. 
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[asterisk-users] Too Many Open Files, Hung SIP Sessions, Can I Increase File Count?

2007-03-21 Thread JR Richardson

Hi All,

 

Something happened on one of my 1.2.9.1 systems, SIP between * and Cisco
Call Manager 4.1, leaving hung or open SIP sessions.  No problem now, we
found and corrected the problem.  But while these hung sessions were
increasing to around 480 to 500 sessions, I started getting too many open
files on the asterisk console and sporadically could not establish new SIP
connections.

 

Now, there was not a dead halt, some SIP sessions could still connect but I
had to restart * to clear out all the hung sessions.

 

I remember something about being able to setup * with increased file count.
Not sure exactly what this refers to but can someone point me in the right
direction?

 

Or am I on the wrong track?

 

Thanks.

 

JR

 

JR Richardson

Engineering for the Masses

 

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Re: [asterisk-users] Too Many Open Files, Hung SIP Sessions, Can I Increase File Count?

2007-03-21 Thread phil . dawson

Here's a snippet taken from the readme.


--

* FILE DESCRIPTORS

  Depending on the size of your system and your configuration,
Asterisk can consume a large number of file descriptors.  In UNIX,
file descriptors are used for more than just files on disk.  File
descriptors are also used for handling network communication
(e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and
digital trunk hardware).  Asterisk accesses many on-disk files for
everything from configuration information to voicemail storage.

  Most systems limit the number of file descriptors that Asterisk can
have open at one time.  This can limit the number of simultaneous
calls that your system can handle.  For example, if the limit is set
at 1024 (a common default value) Asterisk can handle approxiately 150
SIP calls simultaneously.  To change the number of file descriptors
follow the instructions for your system below:

== PAM-based Linux System ==

  If your system uses PAM (Pluggable Authentication Modules) edit
/etc/security/limits.conf.  Add these lines to the bottom of the file:

rootsoftnofile  4096
roothardnofile  8196
asterisksoftnofile  4096
asteriskhardnofile  8196

(adjust the numbers to taste).  You may need to reboot the system for
these changes to take effect.

== Generic UNIX System ==

  If there are no instructions specifically adapted to your system
above you can try adding the command ulimit -n 8192 to the script
that starts Asterisk.


--





Phil.





   
 JR Richardson   
 [EMAIL PROTECTED] 
 mail.com  To 
 Sent by:  asterisk-users@lists.digium.com   
 asterisk-users-bo  cc 
 [EMAIL PROTECTED] 
 m.com Subject 
   [asterisk-users] Too Many Open  
   Files, Hung SIP Sessions, Can I 
 22/03/2007 00:24  Increase File Count?
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




Hi All,

Something happened on one of my 1.2.9.1 systems, SIP between * and Cisco
Call Manager 4.1, leaving hung or open SIP sessions.  No problem now, we
found and corrected the problem.  But while these hung sessions were
increasing to around 480 to 500 sessions, I started getting “too many open
files” on the asterisk console and sporadically could not establish new SIP
connections.

Now, there was not a dead halt, some SIP sessions could still connect but I
had to restart * to clear out all the hung sessions.

I remember something about being able to setup * with increased file count.
Not sure exactly what this refers to but can someone point me in the right
direction?

Or am I on the wrong track?

Thanks.

JR

JR Richardson
Engineering for the Masses
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[asterisk-users] Cisco 30VIP Phone

2007-03-21 Thread Chris Nighswonger


Hi all,
 I have just successfully configured a Cisco 30VIP to work with my
Asterisk server. I have seven of these phones new and would like to
deploy them. I am wondering if anyone has this phone deployed with
Asterisk and can suggest configuration of the various buttons, etc.
(Bare with me as I am new to Asterisk.)

Thanks,
Chris
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Re: [asterisk-users] wrong values in duration and billsec in CDR

2007-03-21 Thread C F


Zap channels on FXO are considered answers as soon as Asterisk finished dialing.

On 3/21/07, Jovanny Saravia [EMAIL PROTECTED] wrote:

Hi to all,

 I was looking in google and also in this mailing list, but I dont find the
solution to my problem, so I subscribe me to the list in order to post this
e-mail and find the solution.

 This is the scenario:

 GSM Phone - GSM Network  TDM2406E ---  ASterisk 1.4.0 (*) 
VoIP Provider --- Sip Phone or H323 Phone

 The problem is that I am generating calls from SIP and also h323 (using
ooh323), and always I saw differences between duration time and billsecs
just for 4 or 3 seconds. Altought the difference is much more, I mean I just
call from SIP and H323 clients and always I saw the same behaviour.

 When I generate the call I wait to pickup the cell phone almost 10 secs,
the right time should be something like 30 secs, but I saw duration = 50,
and billsec = 47. This is a very weird behaviour and I was trying to modify
zaptel.conf but I can't find what my problem is.

 I guess the problem is that this time is being counted just from Voip
domain, and not into the Zaptel domain.

 Maybe some of you could guide me to solve this problem.

 Any help will be so much appreciated

 Rgds,

 Jovanny Saravia
 [EMAIL PROTECTED]

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[asterisk-users] SIP peer disappearing

2007-03-21 Thread Luki


Hi all,

I'm having this weird issue that I can't explain. Maybe someone can
explain what is happening.

This is a Asterisk install that has been in production for 6+ months.
It's version 1.2.10. Couple weeks ago one SIP peer started
disappearing randomly. And I mean it simply disappears. One second
sip show peers shows it, and then it's gone. A simple sip reload
fixes it:

(all good...)

[Mar 21 19:53:53] NOTICE[4481]: chan_sip.c:12049
handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed
for 'IP' - Username/auth name mismatch

lax*CLI sip show peer luki1
Peer luki1 not found.

lax*CLI sip reload
[Mar 21 19:54:10]  Reloading SIP
[Mar 21 19:54:10]   == Parsing '/etc/asterisk/sip.conf': [Mar 21 19:54:10] Found
[Mar 21 19:54:10]   == Parsing '/etc/asterisk/shared/users.conf': [Mar
21 19:54:10] Found
[Mar 21 19:54:10]   == Parsing '/etc/asterisk/sip_notify.conf': [Mar
21 19:54:10] Found

[Mar 21 19:54:22] -- Registered SIP 'luki1' at IP port 5060 expires 60
[Mar 21 19:54:22] -- Saved useragent Linksys/SPA2102-3.3.5(a)
for peer luki1


It only affects one peer. Sometimes it disappears after a few hours,
sometimes after a week. The box can be mostly idle or loaded. No
difference. I did restart Asterisk completely, no help. Upgrading is
an option, but so far there was not need.

Any ideas what is happening?! This peer has been fine for months, and
now this. Line 2 on the Sipura registers with another box (it's
actually Asterisk 1.2.5) -- no problems there. And yes, I did reboot
the Sipura.

--Luki
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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

2007-03-21 Thread Hermann Wecke


Richard Klingler wrote:

Has any1 got their 7970 to work with * 1.4.x ?


Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between them - 
check http://preview.tinyurl.com/345fmj

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Re: [asterisk-users] Refund from SellVoip?

2007-03-21 Thread Ira


At 09:08 AM 3/21/2007, you wrote:

 Does anybody know Jed Stafford?  As far as I can tell this ended up
 being a one-man or two-man operation.  It's just sad.


I got a marketing email from them last week telling me about all 
their cool new features.


Ira 


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Re: [asterisk-users] ses ActiveDirectory and also Ldap and Kerberos.

2007-03-21 Thread dave cantera

carlos,
this is coming from a linux admin perspective but here is something to
get started...
active directory transfers info to-from windows domain controllers via
the network. there are probably api frameworks available as open
source, although they may be incomplete. I have seen another vendor
create a product that when you logged into the windows box, the startup
script had a program that also logged into a linux box...
you would have to write that binary yourself... once you did it on
windows, you would have to port it to linux. I would look to the
samba project as a start... they do windows file sharing and probably
login to widows controllers to get authentication.. haven't used it
extensively though in years...
I know that it is doable, would take a lot of research if you never
used samba... let me google something and see if I can get any more
info...
ok,
you might start here...

http://www.google.com/search?hl=enq=%22active+directory%22+%22single+signon%22+sambabtnG=Google+Search
good luck,
I would be interested to see what you come up with... please repost the
results here!
daveC

Carlos Jernimo wrote:
Hi i'm student and my final project is related to Voip. I
have

Asterisk almost fully configured. The next step is to accept login of

users, that data is in Universitys database which uses ActiveDirectory

and also Ldap and Kerberos.

It's possible? I don't want authentications in sip.conf, but in other

remote database.

The problem is i don't have ideas how to start with.

I would appreciate some ideas

--
Building Strong Relationships w/ Intelligent Customer Service
--

Interlocking Business Solutions, LLC
856-380-0894 x5000

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[asterisk-users] CN=Diarmaid O'Loughlin/O=QAD1 is out of the office.

2007-03-21 Thread Diarmaid O'Loughlin


I will be out of the office starting  22-03-2007 and will not return until
16-04-2007.

I will be on holidays until the 16th of April. I will have no access to
e-mail during this time. If this is a matter relating to IT support, please
contact the IT Helpdesk by either opening a ticket via
http://helpdesk.qad.com/request. Or calling the EMEA helpdesk on +31 20 654
7139.

If this is an urgent matter not relating to IT support. Please contact
either Jim Josey (jzj-at-qad.com), Paul Callan (plc-at-qad.com) or you may
contact me again after Apr 16th.

Thanks,
Diarmaid.


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Re: [asterisk-users] Cisco 30VIP Phone


On 3/21/07, Chris Nighswonger [EMAIL PROTECTED] wrote:


I have just successfully configured a Cisco 30VIP to work with my
Asterisk server. I have seven of these phones new and would like to
deploy them. I am wondering if anyone has this phone deployed with
Asterisk and can suggest configuration of the various buttons, etc.
(Bare with me as I am new to Asterisk.)



So which is it?  You either have it configured or you don't.
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Re: [asterisk-users] Asterisk 1.4.2 chan_zap

2007-03-21 Thread Russell Bryant


Jeremiah Millay wrote:
Am I to understand that to be able to build chan_zap with this new 
version of asterisk I also need to have a new version of zaptel (which 
has yet to be released) that includes ZT_EVENT_REMOVED in zaptel.h? I'm 
able to compile chan_zap under asterisk 1.4.1 against zaptel 1.4.0 just 
fine.


That is exactly correct.  In the meantime, you can use zaptel from svn and it 
will work fine.


svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4

--
Russell Bryant
Software Engineer
Digium, Inc.
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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1


On 3/21/07, Richard Klingler [EMAIL PROTECTED] wrote:


As chan_sccp is pretty much dead, doesn't compile on FBSD anyway
and isn't supported on * 1.4.x I tried going with chan_skinny...



chan_sccp is far from dead and it works with 1.4.  more fud being spread...
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Re: [asterisk-users] Asterisk 1.4.2 chan_zap


On 3/21/07, Jeremiah Millay [EMAIL PROTECTED] wrote:


chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0.
The changelog has this entry:



And you missed all the other discussions about it not working?  Or, are you
just special and wanted your own thread?
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Re: [asterisk-users] A request for your input.


On 3/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


Hello


P.S The program that I am using is open source, of course
(www.phpsurveyor.org)!



What part of the survey is running Asterisk?
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Re: [asterisk-users] CN=Diarmaid O'Loughlin/O=QAD1 is out of the office.