Re: [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall

2010-03-28 Thread Warren Selby 
On Mon, Mar 29, 2010 at 12:25 AM, Troy Davis  wrote:

>
>> sip fixup is enabled on the PIX
>>
>
>
Try disabling the sip fixup on the PIX and see if that helps.  You may have
to adjust the configs on the phones themselves when you do this.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] dnd not working correctly

2010-03-28 Thread Ioan Indreias 
I would say that from what I know DND function in FreePBX will not
automatically configure the phone DND function but it set a flag into
Asterisk DB:

-- Executing [...@from-internal:5] Set("SIP/117-01f6",
"DB(DND/117)=YES") in new stack

You report that you do not hear nothing but in the log we see:

-- Executing [...@from-internal:8] Playback("SIP/117-01f6",
"do-not-disturb&activated") in new stack

I advice to investigate why you are not hearing those prompts before
any other steps.

HTH,
Ioan.

On Sat, Mar 27, 2010 at 12:41 AM, Ott Rose  wrote:
> i have posted this question couple of times and never really got any hits i
> wasn't able to provide any debug info
>

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Re: [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall

2010-03-28 Thread Troy Davis 
>
> I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
> After some period of time, asterisk says that some of them are
> unreachable, and the phones lose their registration.
> The only way to make the phones recover is to clear the NAT
> translation tables for the phones on the PIX (clear xlate...)
> Does anyone know how to fix this? As you can imagine, it is quite
> annoying. And it does not happen to all the phones either.
>
> sip fixup is enabled on the PIX
>

Are you able to TFTP new phone configs?  Assuming so, and it's for only 10
phones, try decreasing the registration time.  I've got a 7960 on my desk
and documented it with a TFTP-ready config:
http://help.cloudvox.com/faqs/sip-phones/cisco-7900-ip-phone

It's at the end, commented out.  I don't think that config's been used much
- most Cloudvox folks are just using SIP to test their AGI apps, not as
primary phones.

If you want another data point that still crosses your NAT boundary, feel
free to sign up for and register with Cloudvox and see whether your
registration lasts, using that same config.  We switched to pay-as-you-go
pricing, so even the free accounts include SIP.  If your registrations to
Cloudvox also time out, it's probably the PIX.

Troy

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Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk

2010-03-28 Thread James Lamanna 
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards
 wrote:
> On Sun, 28 Mar 2010, Joseph Begumisa wrote:
>
>> Can anyone recommend a 24 fxs port voip gateway that has worked well with
>> asterisk?  I have a couple of analog handsets that I want to hookup to my
>> asterisk server?  Any tested and tried product recommendations are welcome.
>>  Thanks.
>
> Adtran channel banks are a great "trailing edge" technology. You can get
> them off Ebay for pennies on the original dollar and they are built like a
> tank.
>
> ("voip gateway" is not very specific. If you meant SIP or IAX, you might
> want to specify which.)

I've actually had decent success with the GXW-4024 (FXS <-> SIP) from
Grandstream which is probably one of the cheapest 24 FXS port boxes
you'll find out there.

-- James

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Re: [asterisk-users] Trying to configure xorcom on Suse 11

2010-03-28 Thread Jim Dickenson 
This is what I get on my unit but my is working:

dahdi_hardware -v
Failed running '/usr/sbin/astribank_tool':  at 
/usr/lib/perl5/site_perl/5.8.8/Dahdi/Xpp/Mpp.pm line 181.
usb:001/003  xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
 LABEL=[usb:X1036340]   connect...@usb-:00:1a.0-1.4
XBUS-00/XPD-00: T1   (24)  Span 1  DAHDI-SYNC
XBUS-00/XPD-10: FXS  (8)   Span 2 
XBUS-00/XPD-20: FXO  (8)   Span 3 

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Mar 28, 2010, at 7:16 PM, Tim Panton wrote:

> 
> On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote:
> 
>> On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
>>> I'm having trouble getting a xorcom set up.
>>> 
>>> A large part of the problem is that the box is a _long_ way away and 
>>> I can't get to/at it easily, so while I could probably fix this in a few
>>> hours if the machine were in front of me, I'm struggling over a slow
>>> unreliable laggy link. 
>>> 
>>> Ok, enough whining from me.
>>> 
>>> I have a new Xorcom plugged into the usb of a Suse 11 machine
>>> I built Dahdi from trunk (last thursday) 
>>> 
>>> # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux
>>> # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools
>>> 
>>> dahdihardware -v sees the box but no spans.
>> 
>> Generally '/etc/init.d/dahdi start' . Or more specifically,
>> 'dahdi_registration on' .
>> 
>> See also:
>> 
>> http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios
> 
> 
> I've must be missing something here - this is what I see now.
> 
> sh-4.0# dahdi_hardware -v
> usb:001/020  xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
> LABEL=[usb:X1037246]   connect...@usb-:00:1a.7-4 
> 
> Shouldn't I see spans ??? I think the box (I've never seen it, but I know 
> what I asked for) 
> has 8fxs+8fxo+2E1 . 
> 
> 
> Thanks, 
> Tim.
> 
> 
> 
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
> 
> 
> 
> 
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Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk

2010-03-28 Thread Steve Edwards 

On Sun, 28 Mar 2010, Joseph Begumisa wrote:

Can anyone recommend a 24 fxs port voip gateway that has worked well 
with asterisk?  I have a couple of analog handsets that I want to hookup 
to my asterisk server?  Any tested and tried product recommendations are 
welcome.  Thanks.


Adtran channel banks are a great "trailing edge" technology. You can get 
them off Ebay for pennies on the original dollar and they are built like a 
tank.


("voip gateway" is not very specific. If you meant SIP or IAX, you might 
want to specify which.)


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000-- 
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Re: [asterisk-users] Trying to configure xorcom on Suse 11

2010-03-28 Thread Tim Panton 

On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote:

> On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
>> I'm having trouble getting a xorcom set up.
>> 
>> A large part of the problem is that the box is a _long_ way away and 
>> I can't get to/at it easily, so while I could probably fix this in a few
>> hours if the machine were in front of me, I'm struggling over a slow
>> unreliable laggy link. 
>> 
>> Ok, enough whining from me.
>> 
>> I have a new Xorcom plugged into the usb of a Suse 11 machine
>> I built Dahdi from trunk (last thursday) 
>> 
>> # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux
>> # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools
>> 
>> dahdihardware -v sees the box but no spans.
> 
> Generally '/etc/init.d/dahdi start' . Or more specifically,
> 'dahdi_registration on' .
> 
> See also:
> 
>  
> http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios


I've must be missing something here - this is what I see now.

sh-4.0# dahdi_hardware -v
usb:001/020  xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
 LABEL=[usb:X1037246]   connect...@usb-:00:1a.7-4 

Shouldn't I see spans ??? I think the box (I've never seen it, but I know what 
I asked for) 
has 8fxs+8fxo+2E1 . 


Thanks, 
Tim.



Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




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Re: [asterisk-users] Cisco 7960 become UNREACHABLE behind pix firewall

2010-03-28 Thread James Lamanna 
Alyed wrote:

> From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
> "If you turn on *qualify* in the configuration of a SIP device in
> sip.conf,
> asterisk will send a SIP
> OPTIONScommand
> regularly to check that the device is still online. If the device
> does not answer within the configured (or default) period (in ms) Asterisk
> considers the device off-line for future calls. This status can be checked
> by the SIPPEER 
> function,
> and inversely this function will only provide status information for peers
> which have *qualify=yes*."
> My guess is that your Nat/firewall is closing the connection after some time
> the phone is idle, so this way Asterisk will make sure to always have
> communication going trhough that connection so your NAT/firewall won't just
> close it.

Sorry, should have mentioned that all these phones have qualify=yes
and nat=yes in sip.conf.

Thanks.

-- James

> On Sat, Mar 27, 2010 at 8:17 AM, James Lamanna  wrote:
>> Hi,
>> I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
>> After some period of time, asterisk says that some of them are
>> unreachable, and the phones lose their registration.
>> The only way to make the phones recover is to clear the NAT
>> translation tables for the phones on the PIX (clear xlate...)
>> Does anyone know how to fix this? As you can imagine, it is quite
>> annoying. And it does not happen to all the phones either.
>>
>> sip fixup is enabled on the PIX
>>
>> phone config parts:
>>
>> nat_enable : 1
>> nat_received_processing : 0
>> nat_address: [public ip of PIX]

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Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk

2010-03-28 Thread Doug 
At 19:55 3/28/2010, Joseph Begumisa wrote:
>Hi,
>
>Can anyone recommend a 24 fxs port voip gateway 
>that has worked well with asterisk? Â I have a 
>couple of analog handsets that I want to hookup 
>to my asterisk server? Â Any tested and tried 
>product recommendations are welcome. Â Thanks.

You may want to buy 12 Linksys PAP2s.  Have used
8 port, and 24 port boxes in the past.  The problem
with this is if just one port goes bad, you need
to replace the whole unit.  Also, you need to keep
spares on hand.

 From getting burned in the past, I would recommend
using multiples of these 2 port ATAs, and keeping a
few spares on hand.






>Best Regards,
>
>Joseph
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Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer

2010-03-28 Thread kamrun nahar bina 
Dear sir,

Thanks for your reply. We have tested in another phone like Bria(2.4.3 buid
50906) with same phenomenon. But we are getting same error "Failed to play
transfer sound! " during attended transfer.

Is there anything which causes this problem? And we are not facing this
problem first time. Before we faced in this problem occasionally. But
recently, this problem occurs frequently.

Is there any other problem or any other prerequisite for this problem? Or is
it the problem of asterisk?  How we can overcome this problem ?
Please give us solution.

Thanks in advance

Nahar




On Sat, Mar 27, 2010 at 1:33 AM, Alyed  wrote:

> so doesn't looks like overload
>
> Could it be a problem with the firmware of your softphones? Have you been
> using some new phones lately? someone else in a different thread pointed on
> attended transfer bugs with SNOM phones.
>
>
> > We are eagerly waiting for your solution.
> Hope we can help but don't so much pressure on me or the listers :)
>
> Alyed
>
>
>
> 2010/3/26 kamrun nahar bina 
>
> Dear sir,
>>
>> Thanks for your reply.
>>
>> our memory size is 4GB.
>> concurrent calls no : 30.
>> Our memory condition is below :
>>
>> Cpu(s):  0.3%us,  0.7%sy,  0.0%ni, 98.5%id,  0.0%wa,  0.1%hi,  0.3%si,
>> 0.0%st
>> Mem:   4147888k total,  3986540k used,   161348k free,76852k buffers
>> Swap:  2031608k total,   56k used,  2031552k free,  3170396k cached
>>
>>   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
>> 23160 root  15   0  440m 415m 5688 S  4.3 10.3 398:13.93 asterisk
>>
>> Our disk space condition is below:
>> FilesystemSize  Used Avail Use% Mounted on
>> /dev/mapper/VolGroup00-LogVol00
>>   901G  245G  610G  29% /
>> /dev/sda1  99M   18M   77M  19% /boot
>> tmpfs 2.0G 0  2.0G   0% /dev/shm
>>
>>
>> We are eagerly waiting for your solution.
>>
>> Thanks in advance.
>>
>> Nahar
>>
>>
>>
>> On Fri, Mar 26, 2010 at 2:32 PM, Alyed  wrote:
>>
>>> If you didn't have this problem before I'll check up for any changes
>>> lately (i suppose you have done so, but ask this just to be safe)
>>> I see you have lots of agents and also lots of hard disk space, so I
>>> guess disk space is not an issue. Please check it anyway.
>>>
>>> how many concurrent calls you have? 2 GB in RAM seems little against 600
>>> registered agents.
>>>
>>> Alyed
>>>
>>>
>>> 2010/3/25 kamrun nahar bina 
>>>
 Dear sir,

 We have been using asterisk for 4 years. Now we have got problems which
 occurs during the attended transfer.
 But we are not always getting this problem. Sometimes it happens. But
 now we cannot understand why this is happening?

 problem is:"Failed to play transfer sound! "

 The log of asterisk is as like as followings:

 [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP message -
 rejected , no callid, len 366




 [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was
 pretty quick last time, waiting for them.
 [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was
 pretty quick last time, waiting for them.




 [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on
 dialog 5bd1acee539e699b4f5e79c94a348...@113.34.235.8

 [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Received bye, issuing owner
 hangup
 [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was
 pretty quick last time, waiting for them.
 [Mar 25 17:58:40] WARNING[13591] res_features.c: Failed to play transfer




 sound!

 Our system is as like as:
 The number of User agent is: 1650
 The number of Actual registered user agent is: 600

 Our System configuration is :
 IBM X3550
 CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz




 Memory: 2GB
 HDD: 3.5 SATA 1TB x 2
 version of asterisk: 1.4.23.1

 Asterisk and the User-Agent is connected through the Internet.




 ..And Is there any solution to solve this problem? We have 
 investigated in several places but we cannot find out the reason?
 We need this solution very urgently. We are eagerly waiting for reply.

 Thanks in advance




 Nahar


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Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk

2010-03-28 Thread Zeeshan Zakaria 
I have used ones from Rhino and no complains.

On 2010-03-28 9:00 PM, "Joseph Begumisa"  wrote:

Hi,

Can anyone recommend a 24 fxs port voip gateway that has worked well with
asterisk?  I have a couple of analog handsets that I want to hookup to my
asterisk server?  Any tested and tried product recommendations are welcome.
 Thanks.

Best Regards,

Joseph

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[asterisk-users] 24 FXS Port Voip Gateway and Asterisk

2010-03-28 Thread Joseph Begumisa 
Hi,

Can anyone recommend a 24 fxs port voip gateway that has worked well with
asterisk?  I have a couple of analog handsets that I want to hookup to my
asterisk server?  Any tested and tried product recommendations are welcome.
 Thanks.

Best Regards,

Joseph
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Re: [asterisk-users] SIP/2.0 403 Forbidden

2010-03-28 Thread Dovid Bender 
You need to ask your carrier what you are not sending them that they would 
like. It's usually a fromdomain or authname.
  - Original Message - 
  From: Aaron chen 
  To: Asterisk Users Mailing List - Non-Commercial Discussion ; Asterisk 
Developers Mailing List 
  Sent: Friday, March 26, 2010 09:22
  Subject: [asterisk-users] SIP/2.0 403 Forbidden


  hi,all

  when i send a call to other server use SIP trunk,

  i got this below,

  <--- SIP read from 222.46.18.52:5060 --->
  SIP/2.0 403 Forbidden

  what's rong with is?


Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it 
under
certain conditions. Type 'core show license' for details.
=
  == Parsing '/etc/asterisk/asterisk.conf': Found
Connected to Asterisk 1.4.21.2 currently running on gd-branch (pid = 3145)
Verbosity is at least 3
-- Executing [015921256...@from-internal:1] Set("SIP/75002-b7705298", 
"MOHCLASS=none") in new stack
-- Executing [015921256...@from-internal:2] Macro("SIP/75002-b7705298", 
"user-callerid|SKIPTTL|") in new stack
-- Executing [...@macro-user-callerid:1] Set("SIP/75002-b7705298", 
"AMPUSER=75002") in new stack
-- Executing [...@macro-user-callerid:2] GotoIf("SIP/75002-b7705298", 
"0?report") in new stack
-- Executing [...@macro-user-callerid:3] ExecIf("SIP/75002-b7705298", 
"1|Set|REALCALLERIDNUM=75002") in new stack
-- Executing [...@macro-user-callerid:4] Set("SIP/75002-b7705298", 
"AMPUSER=75002") in new stack
-- Executing [...@macro-user-callerid:5] Set("SIP/75002-b7705298", 
"AMPUSERCIDNAME=75002") in new stack
-- Executing [...@macro-user-callerid:6] GotoIf("SIP/75002-b7705298", 
"0?report") in new stack
-- Executing [...@macro-user-callerid:7] Set("SIP/75002-b7705298", 
"AMPUSERCID=75002") in new stack
-- Executing [...@macro-user-callerid:8] Set("SIP/75002-b7705298", 
"CALLERID(all)="75002" <75002>") in new stack
-- Executing [...@macro-user-callerid:9] ExecIf("SIP/75002-b7705298", 
"0|Set|CHANNEL(language)=") in new stack
-- Executing [...@macro-user-callerid:10] GotoIf("SIP/75002-b7705298", 
"1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [...@macro-user-callerid:19] NoOp("SIP/75002-b7705298", 
"Using CallerID "75002" <75002>") in new stack
-- Executing [015921256...@from-internal:3] Set("SIP/75002-b7705298", 
"_NODEST=") in new stack
-- Executing [015921256...@from-internal:4] Macro("SIP/75002-b7705298", 
"record-enable|75002|OUT|") in new stack
-- Executing [...@macro-record-enable:1] GotoIf("SIP/75002-b7705298", 
"1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI("SIP/75002-b7705298", 
"recordingcheck|20100326-141638|1269584198.62") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20100326-141638|1269584198.62: Outbound recording enabled.
  recordingcheck|20100326-141638|1269584198.62: 
CALLFILENAME=OUT75002-20100326-141638-1269584198.62
-- AGI Script recordingcheck completed, returning 0
-- Executing [...@macro-record-enable:999] 
MixMonitor("SIP/75002-b7705298", 
"/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav||")
 in new stack
-- Executing [...@macro-record-enable:1000] Set("SIP/75002-b7705298", 
"RecordingFileName=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav")
 in new stack
-- Executing [...@macro-record-enable:1001] NoOp("SIP/75002-b7705298", 
"/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav")
 in new stack
-- Executing [...@macro-record-enable:1002] Set("SIP/75002-b7705298", 
"CDR(userfield)=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav")
 in new stack
-- Executing [015921256...@from-internal:5] Macro("SIP/75002-b7705298", 
"dialout-trunk|7|015921256331||") in new stack
-- Executing [...@macro-dialout-trunk:1] Set("SIP/75002-b7705298", 
"DIAL_TRUNK=7") in new stack
-- Executing [...@macro-dialout-trunk:2] GosubIf("SIP/75002-b7705298", 
"0?sub-pincheck|s|1") in new stack
-- Executing [...@macro-dialout-trunk:3] GotoIf("SIP/75002-b7705298", 
"0?disabletrunk|1") in new stack
-- Executing [...@macro-dialout-trunk:4] Set("SIP/75002-b7705298", 
"DIAL_NUMBER=015921256331") in new stack
-- Executing [...@macro-dialout-trunk:5] Set("SIP/75002-b7705298", 
"DIAL_TRUNK_OPTIONS=Ttr") in ne

Re: [asterisk-users] Libtonezone

2010-03-28 Thread Anthony Francis - Handy Networks LLC 
You could read the source code, but based on it's name I would say it is a 
library responsible for zone specific tone generation. Many parts of the world 
have different tone patterns than the U.S. and Asterisk is used worldwide. A 
better question is, why are you concerned by it?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] on behalf of Joseph L. Casale 
[jcas...@activenetwerx.com]
Sent: Sunday, March 28, 2010 9:13 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Libtonezone

Trying to find out what the libtonezone shared object built with dahdi-tools is
for, the default dahdi package installation from the Digium repo's pull it in,
so when is it needed?

Thanks,
jlc

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Re: [asterisk-users] Updating Asterisk and its use with MySQL

2010-03-28 Thread Alyed 
Yes I'm talking about Asterisk Now's GUI and yes, you can just install this
component.
google for Asterisk Gui 2.0 and you'll find plenty of info.

Regarding the DB I can't help you here, maybe someone else can.

Alyed


2010/3/28 Daniel Bareiro 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On Sun, 28 Mar 2010, Alyed wrote:
>
> >> My idea is to continue making the configurations by hand at the
> >> moment, that it is the way that I used until now, to familiarize to
> >> me with the handling of Asterisk at lower level, without using a
> >> graphical interface, and in a later stage of the tests to take these
> >> configurations through something like FreePBX. What think of this
> >> form to think?
>
> > I would suggest trying Digium's GUI first and then FreePBX since the
> > first one I find it more readable. You'll find out eventually that
> > there's no easy way to migrate from pure command line to a GUI, but
> > you'll learn a lot in the meantime.
>
> I didn't know that there was Digium's GUI. It is FLOSS? I was looking
> for in the site of Digium in the download section, but the unique thing
> that I saw that it speaks of a GUI is AsteriskNow, that in fact it is a
> complete distribution of GNU/Linux. You talked about to the GUI provided
> by AsteriskNow? Because if is this case, I don't believe that it is very
> practical. When I spoke of GUI was referring to a separated component to
> install over which already one had running.
>
> As far as the use of Asterisk with a DBMS (MySQL, for example), do you
> know some document or reference where indicate the steps to follow to
> migrate from config files?
>
> Thanks for your reply.
>
> Regards,
> Daniel
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (GNU/Linux)
>
> iEYEARECAAYFAkuvu8oACgkQZpa/GxTmHTdFVACePM0WaIfeHQmM+w8cpLuGGt/5
> XSAAoI+YrC+9Y91ElRhFBrxAG6XVxEyh
> =e4iN
> -END PGP SIGNATURE-
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Updating Asterisk and its use with MySQL

2010-03-28 Thread Daniel Bareiro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Sun, 28 Mar 2010, Alyed wrote:

>> My idea is to continue making the configurations by hand at the
>> moment, that it is the way that I used until now, to familiarize to
>> me with the handling of Asterisk at lower level, without using a
>> graphical interface, and in a later stage of the tests to take these
>> configurations through something like FreePBX. What think of this
>> form to think?

> I would suggest trying Digium's GUI first and then FreePBX since the
> first one I find it more readable. You'll find out eventually that
> there's no easy way to migrate from pure command line to a GUI, but
> you'll learn a lot in the meantime.

I didn't know that there was Digium's GUI. It is FLOSS? I was looking
for in the site of Digium in the download section, but the unique thing
that I saw that it speaks of a GUI is AsteriskNow, that in fact it is a
complete distribution of GNU/Linux. You talked about to the GUI provided
by AsteriskNow? Because if is this case, I don't believe that it is very
practical. When I spoke of GUI was referring to a separated component to
install over which already one had running.

As far as the use of Asterisk with a DBMS (MySQL, for example), do you
know some document or reference where indicate the steps to follow to
migrate from config files?

Thanks for your reply.

Regards,
Daniel

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=e4iN
-END PGP SIGNATURE-


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Re: [asterisk-users] Updating Asterisk and its use with MySQL

2010-03-28 Thread Alyed 
>My idea is to continue making the configurations by hand at the moment,
>that it is the way that I used until now, to familiarize to me with the
>handling of Asterisk at lower level, without using a graphical
>interface, and in a later stage of the tests to take these
>configurations through something like FreePBX. What think of this form
>to think?

I would suggest trying Digium's GUI first and then FreePBX since the first
one I find it more readable. You'll find out eventually that there's no easy
way to migrate from pure command line to a GUI, but you'll learn a lot in
the meantime.

Have Fun!

Alyed



2010/3/28 Daniel Bareiro 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> - -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi, Jim.
>
> On Sun, 28 Mar 2010, Jim Dickenson wrote:
>
> >>> Make sure not to do "make samples" or you will overwrite your .conf
> >>> file. This is the important one to watch out for. You can save off
> >>> your .conf files and then restore them or compare your files with
> >>> the new ones to see if there are any important new settings.
>
> >> I had thought that "make config" was what I would have to avoid.
> >> Which is the difference? does "make config" create the init scripts
> >> and "make samples" the example configuration files?
>
> > Yes, "make config" installs /etc/init.d/asterisk on Linux systems and
> > does the appropriate chkconfig steps so will start on boot while "make
> > samples" installs the .conf files in, by default, /etc/asterisk.
>
> Perfect.
>
> >> Do these two "makes" have the same behavior for Asterisk and DAHDI? I
> >> have understood that "make config" in DAHDI Tools is the one that
> >> creates both the configuration files and init scripts.
>
> > There is no "make config" for dahdi. I think /etc/dahdi files do not
> > get overwritten if they are there already.
>
> Hmmm... nevertheless I have documented this procedure in my Dokuwiki of
> the time that I made the installation and compilation:
>
> # tar xvzf dahdi-linux-2.1.0.4.tar.gz
> # tar xvzf dahdi-tools-2.1.0.2.tar.gz
>
> ~/Asterisk/dahdi-linux-2.1.0.4# make
> ~/Asterisk/dahdi-linux-2.1.0.4# make install
>
> ~/Asterisk/dahdi-tools-2.1.0.2# ./configure
> ~/Asterisk/dahdi-tools-2.1.0.2# make menuselect   # In order to select a
> customized configuration
> ~/Asterisk/dahdi-tools-2.1.0.2# make
> ~/Asterisk/dahdi-tools-2.1.0.2# make install
> ~/Asterisk/dahdi-tools-2.1.0.2# make config   # In order to install
> scripts and config files
>
> >> When I compiled the version that I'm using at the moment of DAHDI
> >> Linux only I used "make" and "make install" without using "make
> >> samples" or "make config". Are also generated configuration files
> >> with DAHDI Linux?
>
> > I think if you are installing dahdi complete from source you do "make
> > all" and "make install" and "make config"
>
> Thanks. I will consider it if I install this package of DAHDI.
>
>
> Thanks for your reply.
>
> Regards,
> Daniel
>
> - -BEGIN PGP SIGNATURE-
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>
> iEYEARECAAYFAkuvqhAACgkQZpa/GxTmHTfqMQCfT2V7RR4JMFp/EpH4J0F8Tfk9
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> =Q6s+
> - -END PGP SIGNATURE-
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>
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> 4QQAnjWRKkOJls9dFVwVM0IQORkmDIPd
> =YxoR
> -END PGP SIGNATURE-
>
>
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> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Updating Asterisk and its use with MySQL

2010-03-28 Thread Daniel Bareiro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

- -BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi, Jim.

On Sun, 28 Mar 2010, Jim Dickenson wrote:

>>> Make sure not to do "make samples" or you will overwrite your .conf
>>> file. This is the important one to watch out for. You can save off
>>> your .conf files and then restore them or compare your files with
>>> the new ones to see if there are any important new settings.

>> I had thought that "make config" was what I would have to avoid.
>> Which is the difference? does "make config" create the init scripts
>> and "make samples" the example configuration files?

> Yes, "make config" installs /etc/init.d/asterisk on Linux systems and
> does the appropriate chkconfig steps so will start on boot while "make
> samples" installs the .conf files in, by default, /etc/asterisk.

Perfect.

>> Do these two "makes" have the same behavior for Asterisk and DAHDI? I
>> have understood that "make config" in DAHDI Tools is the one that
>> creates both the configuration files and init scripts.

> There is no "make config" for dahdi. I think /etc/dahdi files do not
> get overwritten if they are there already.

Hmmm... nevertheless I have documented this procedure in my Dokuwiki of
the time that I made the installation and compilation:

# tar xvzf dahdi-linux-2.1.0.4.tar.gz
# tar xvzf dahdi-tools-2.1.0.2.tar.gz

~/Asterisk/dahdi-linux-2.1.0.4# make
~/Asterisk/dahdi-linux-2.1.0.4# make install

~/Asterisk/dahdi-tools-2.1.0.2# ./configure
~/Asterisk/dahdi-tools-2.1.0.2# make menuselect   # In order to select a 
customized configuration
~/Asterisk/dahdi-tools-2.1.0.2# make
~/Asterisk/dahdi-tools-2.1.0.2# make install
~/Asterisk/dahdi-tools-2.1.0.2# make config   # In order to install scripts 
and config files

>> When I compiled the version that I'm using at the moment of DAHDI
>> Linux only I used "make" and "make install" without using "make
>> samples" or "make config". Are also generated configuration files
>> with DAHDI Linux?

> I think if you are installing dahdi complete from source you do "make
> all" and "make install" and "make config"

Thanks. I will consider it if I install this package of DAHDI.


Thanks for your reply.

Regards,
Daniel

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=YxoR
-END PGP SIGNATURE-


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[asterisk-users] Trying to configure xorcom on Suse 11

2010-03-28 Thread JD Austin 
I've never seen e4e4:1164 before.

What does this output?:
lsusb|sed -e 's/:/ /g'| grep e4e4| awk '{print "astribank_tool -n -D
/proc/bus/usb/"$2"/"$4}'| bash
reset the astribank:
#(if you use freepbx)
amportal stop
#(if you start asterisk that way)
/etc/init.d/asterisk stop
/etc/init.d/dahdi stop
/usr/share/dahdi/xpp_fxloader reset
#give it time
sleep 30
/usr/share/dhadi/xpp_fxloader load
#(you should see e4e4:1162)
lsusb
#(you should see the hardware here)
dahdi_hardware -v
#presuming you have /etc/dahdi/system.conf right this will work
/etc/init.d/dahdi start
#if you use freepbx
amportal start
#or
/etc/init.d/asterisk start

-- 
JD Austin
Twin Geckos Technology Services LLC
j...@twingeckos.com
Voice: 480.288.8195x201
Fax: 480.406.6753
http://www.twingeckos.com
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Re: [asterisk-users] problem with polarity reverse

2010-03-28 Thread Tzafrir Cohen 
On Sat, Mar 27, 2010 at 08:02:02PM +0200, Justas Gulbinskas wrote:
> Hi,
> 
> I have a problem with polarity reverse 
> I use asterisk 1.4.30 linux kernel version 2.6.27 dahdi version 2.2.1 and 
> analog card is Sangoma a400 with fxo ports
> 
> this is my config 
>   
>   
> [trunkgroups] 
>   
>   
> 
> [channels]
>   
>   
> context=default   
>   
>   
> usecallerid=yes   
>   
>   
> hidecallerid=no   
>   
>   
> callwaiting=yes   
>   
>   
> usecallingpres=yes
>   
>   
> callwaitingcallerid=yes   
>   
>   
> threewaycalling=yes   
>   
>   
> transfer=yes  
>   
>   
> canpark=yes   
>   
>   
> cancallforward=yes
>   
>   
> callreturn=yes
>   
>   
> echocancel=yes
>   
>   
> echocancelwhenbridged=yes 
>   
>   
> relaxdtmf=yes 
>   
>   
> rxgain=0.0
>   
>   
> txgain=0.0
>   
>   
> group=1   
>   
>   
> callgroup=1   
>   
>   
> pickupgroup=1 
>   
>   
> immediate=no  
>   
>   
> answeronpolarityswitch=yes
>   
>   
> 
> 
> 
> and then i call from sip to mobile over gsm gw (nokia 32) which have a 
> polarity reverse i pick up the mobile phone  in sip phone i hear that 
> polarity revers was but at the in asterisk  
> 
> core show channels verbose
> 
> Channel

Re: [asterisk-users] Updating Asterisk and its use with MySQL

2010-03-28 Thread Jim Dickenson 

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Mar 28, 2010, at 10:19 AM, Daniel Bareiro wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Hi, Jim.
> 
> On Sun, 28 Mar 2010, Jim Dickenson wrote:
> 
>> Make sure not to do "make samples" or you will overwrite your .conf
>> file. This is the important one to watch out for. You can save off
>> your .conf files and then restore them or compare your files with the
>> new ones to see if there are any important new settings.
> 
> I had thought that "make config" was what I would have to avoid. Which
> is the difference? does "make config" create the init scripts and "make
> samples" the example configuration files?

Yes, "make config" installs /etc/init.d/asterisk on Linux systems and
does the appropriate chkconfig steps so will start on boot while
"make samples" installs the .conf files in, by default, /etc/asterisk.

> 
> Do these two "makes" have the same behavior for Asterisk and DAHDI? I
> have understood that "make config" in DAHDI Tools is the one that
> creates both the configuration files and init scripts.

There is no "make config" for dahdi. I think /etc/dahdi files do not
get overwritten if they are there already.

> 
> When I compiled the version that I'm using at the moment of DAHDI Linux
> only I used "make" and "make install" without using "make samples" or
> "make config". Are also generated configuration files with DAHDI Linux?
> 

I think if you are installing dahdi complete from source you do
"make all" and "make install" and "make config"

> Thanks for your reply.
> 
> Regards,
> Daniel
> 
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (GNU/Linux)
> 
> iEYEARECAAYFAkuvjioACgkQZpa/GxTmHTcNIgCfZ1PEUqz/3kjGRTa0ECO97jSH
> 53YAni5ICLJGEL2U1Hcwc2hKsDUMYH6V
> =Dp0r
> -END PGP SIGNATURE-
> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
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>   http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] Updating Asterisk and its use with MySQL

2010-03-28 Thread Daniel Bareiro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi, Jim.

On Sun, 28 Mar 2010, Jim Dickenson wrote:

> Make sure not to do "make samples" or you will overwrite your .conf
> file. This is the important one to watch out for. You can save off
> your .conf files and then restore them or compare your files with the
> new ones to see if there are any important new settings.

I had thought that "make config" was what I would have to avoid. Which
is the difference? does "make config" create the init scripts and "make
samples" the example configuration files?

Do these two "makes" have the same behavior for Asterisk and DAHDI? I
have understood that "make config" in DAHDI Tools is the one that
creates both the configuration files and init scripts.

When I compiled the version that I'm using at the moment of DAHDI Linux
only I used "make" and "make install" without using "make samples" or
"make config". Are also generated configuration files with DAHDI Linux?

Thanks for your reply.

Regards,
Daniel

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (GNU/Linux)

iEYEARECAAYFAkuvjioACgkQZpa/GxTmHTcNIgCfZ1PEUqz/3kjGRTa0ECO97jSH
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=Dp0r
-END PGP SIGNATURE-


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Re: [asterisk-users] Updating Asterisk and its use with MySQL

2010-03-28 Thread Jim Dickenson 
Make sure not to do "make samples" or you will overwrite your .conf file. This 
is the important one to watch out for. You can save off your .conf files and 
then restore them or compare your files with the new ones to see if there are 
any important new settings.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Mar 28, 2010, at 8:19 AM, Daniel Bareiro wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Hi all!
> 
> I'm using Asterisk 1.4.24.1 with dahdi-linux-2.1.0.4 and
> dahdi-tools-2.1.0.2 compiled by myself with the source code of the
> official site of the project. I would like to update to one more newer
> version. I suppose that the recommendable thing is to maintain me in
> branch 1.4, reason why in this case it would be 1.4.30 that I suppose
> that it will have several bugs fixed.
> 
> Also I see that there are new versions of DADHI Linux and DAHDI Tools;
> 2.2.1.1 for both cases. I image DAHDI Complete package include both
> DAHDI Linux an DAHDI tools. For this package it is necessary to continue
> making the compilation separately?
> 
> But going to the question to that I make mention in subject, which would
> be the procedure to update the versions of these software maintaining
> the configurations? It is correct to think that the procedure would be
> to stop the Asterisk server and DAHDI, and to follow the same steps for
> the compilation and installation but without doing "make config"?
> 
> On the other hand, at this moment I'm testing with few extensiones on
> low scale, but my idea is to raise the test a little more 50 extensions.
> For this case I suppose that it is more efficient to work with a
> database management system (MySQL, for example) for the configurations
> instead of files. There is some procedure that can recommend to me to
> migrate the configurations in files to a DBMS?
> 
> My idea is to continue making the configurations by hand at the moment,
> that it is the way that I used until now, to familiarize to me with the
> handling of Asterisk at lower level, without using a graphical
> interface, and in a later stage of the tests to take these
> configurations through something like FreePBX. What think of this form
> to think?
> 
> Thanks in advance for your reply and recommendations.
> 
> Regards,
> Daniel
> 
> -BEGIN PGP SIGNATURE-
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> 
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> 
> 
> -- 
> _
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[asterisk-users] Libtonezone

2010-03-28 Thread Joseph L. Casale 
Trying to find out what the libtonezone shared object built with dahdi-tools is
for, the default dahdi package installation from the Digium repo's pull it in,
so when is it needed?

Thanks,
jlc

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[asterisk-users] Updating Asterisk and its use with MySQL

2010-03-28 Thread Daniel Bareiro 
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi all!

I'm using Asterisk 1.4.24.1 with dahdi-linux-2.1.0.4 and
dahdi-tools-2.1.0.2 compiled by myself with the source code of the
official site of the project. I would like to update to one more newer
version. I suppose that the recommendable thing is to maintain me in
branch 1.4, reason why in this case it would be 1.4.30 that I suppose
that it will have several bugs fixed.

Also I see that there are new versions of DADHI Linux and DAHDI Tools;
2.2.1.1 for both cases. I image DAHDI Complete package include both
DAHDI Linux an DAHDI tools. For this package it is necessary to continue
making the compilation separately?

But going to the question to that I make mention in subject, which would
be the procedure to update the versions of these software maintaining
the configurations? It is correct to think that the procedure would be
to stop the Asterisk server and DAHDI, and to follow the same steps for
the compilation and installation but without doing "make config"?

On the other hand, at this moment I'm testing with few extensiones on
low scale, but my idea is to raise the test a little more 50 extensions.
For this case I suppose that it is more efficient to work with a
database management system (MySQL, for example) for the configurations
instead of files. There is some procedure that can recommend to me to
migrate the configurations in files to a DBMS?

My idea is to continue making the configurations by hand at the moment,
that it is the way that I used until now, to familiarize to me with the
handling of Asterisk at lower level, without using a graphical
interface, and in a later stage of the tests to take these
configurations through something like FreePBX. What think of this form
to think?

Thanks in advance for your reply and recommendations.

Regards,
Daniel

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=EKwq
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Re: [asterisk-users] Trying to configure xorcom on Suse 11

2010-03-28 Thread Tzafrir Cohen 
On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
> I'm having trouble getting a xorcom set up.
> 
> A large part of the problem is that the box is a _long_ way away and 
> I can't get to/at it easily, so while I could probably fix this in a few
> hours if the machine were in front of me, I'm struggling over a slow
> unreliable laggy link. 
> 
> Ok, enough whining from me.
> 
> I have a new Xorcom plugged into the usb of a Suse 11 machine
> I built Dahdi from trunk (last thursday) 
> 
> # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux
> # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools
> 
> dahdihardware -v sees the box but no spans.

Generally '/etc/init.d/dahdi start' . Or more specifically,
'dahdi_registration on' .

See also:

  
http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones

2010-03-28 Thread Loic Didelot 
Hi Kevin,
is this feature implemented in the Business Version of Asterisk?

Best reards,
Loïc.



On Thu, 2010-03-25 at 07:20 -0500, Kevin P. Fleming wrote:
> Loic Didelot wrote:
> 
> > I am testing the Openstage phones from Siemens but I can not find a
> > solution on how to update the caller-id after a successful attended
> > transfer. Of course, I mean an attended transfer by using the phones
> > functionality, not something defined in asterisks features.conf.
> 
> This is called 'Connected Party ID', and it isn't supported in any
> released version of Asterisk... but it is supported in SVN trunk and
> will be part of Asterisk 1.8.
> 
> -- 
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
> 



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Re: [asterisk-users] Trying to configure xorcom on Suse 11

2010-03-28 Thread Tim Panton 

On 27 Mar 2010, at 21:48, JD Austin wrote:

> Xorcom hardware uses three layers; you must resolve issues in the 
> following order:
> 
>   1. USB
>   2. Dahdi
>   3. Asterisk
> 
> I suspect you're having trouble with the usb layer.
> Run lsusb
> It will display a line like this if the firmware isn't loaded:
> Bus 001 Device 004: ID e4e4:1161
> If it is e4e4:1162 then the firmware is loaded.
> You can manually load the firmware like this:
> 
>/usr/share/dahdi/xpp_fxloader load
>or
>/usr/share/dahdi/xpp_fxloader usb 
> 


It seems to load (some) usb firmware ok, as you can see from the
syslog, but I suspect it is loading the wrong version.
I got e4e4:1164 (I think - I've lost contact with the box for the moment).

Thanks for the explanation too.

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




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