[asterisk-users] How to test BRI lines energy saving mode ?

2010-10-06 Thread Olivier
Hello,

If my understanding is correct, these days it seems that many ISDN BRI lines
are configured in energy saving mode in which signalling D-channel is
dropped until a new call comes in.

Is it possible to replicate this behaviour with Asterisk (when Asterisk is
in NT mode and is seen as a public ISDN by another PBX, for instance) ?
If not, would you it would be a useful addition to Asterisk ?

Regards
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[asterisk-users] Which virtualization tech to get PCI assignment ?

2010-10-06 Thread Olivier
Hello,

I'm looking for a virtualization technique with which I could easily assign
PCI/PCIe boards to virtual machines.

If this matters, I don't need to be able to use several boards nor to run
several virtual machines at the same time as I'm just looking for a way to
easily mimic several production machines on the same hardware and switch
from one project to another.
In every project, I'm using the same OS (Debian).

In my previous searches, I've read LXC could be matching my needs but I
couldn't read anything about PCI assignment.

Suggestions ?

Regards
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Re: [asterisk-users] Which virtualization tech to get PCI assignment ?

2010-10-06 Thread Thorolf Godawa
Hi,

 I'm looking for a virtualization technique with which I could easily
 assign PCI/PCIe boards to virtual machines.
you might try opensource Xen with paravirtualized Linux-guests, which
supports PCI-passthrough quite good.

Running several Asterisk-servers at one time might be problematic,
because of real time and latency problems, but using just one system
should be possible.
-- 

Chau y hasta luego,

Thorolf

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Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware

2010-10-06 Thread Olivier
2010/10/5 Andrew Latham lath...@gmail.com

 http://www.ip-phone-forum.de/showthread.php?t=188877


For those not fluent in german, what is this thread telling ?



 ~
 Andrew lathama Latham
 lath...@gmail.com

 * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
 * Learn more about Linux http://en.wikipedia.org/wiki/Linux
 * Learn more about Tux http://en.wikipedia.org/wiki/Tux



 On Tue, Jul 3, 2007 at 1:24 PM, Olivier oza-4...@myamail.com wrote:
  Any reply ?
 
  2007/7/1, Olivier oza-4...@myamail.com:
 
  Thanks everybody for your input.
 
  Let me summarize localization process :
 
  1. Buring boot, phones download from TFTP server an xml or older .cfg
 file
  in which a localization parameter is set.
  2. When this parameter is read, phones will then ask CCM or Asterisk or
  another server to send localized button templates using SIP messages.
  3. When SIP messages are received, phones will then localize buttons and
  menus.
  4. Whenever an incoming or outgoing call is processed, phones will no
  further ask any localization data from CCM or Asterisk or anything as
  localization was fully done during initialization process.
 
  A. Is this correct ?
  Specifically, shall I understand that if present initialization process
 is
  extended to include localization, there will be no need to change
 Asterisk
  SIP channel to fully support localized Cisco phones ?
 
  B. What are the SIP messages that trigger and reply to localization
  demands ?
  Is it possible to tailor dialplan so that these Cisco specific messages
  are treated without modifying chan-sip ?
 
  Cheers
 
 
 
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Re: [asterisk-users] Which virtualization tech to get PCI assignment ?

2010-10-06 Thread Gordon Henderson
On Wed, 6 Oct 2010, Olivier wrote:

 Hello,

 I'm looking for a virtualization technique with which I could easily assign
 PCI/PCIe boards to virtual machines.

 If this matters, I don't need to be able to use several boards nor to run
 several virtual machines at the same time as I'm just looking for a way to
 easily mimic several production machines on the same hardware and switch
 from one project to another.
 In every project, I'm using the same OS (Debian).

 In my previous searches, I've read LXC could be matching my needs but I
 couldn't read anything about PCI assignment.

LXC works by having one kernel and many containers kept separate from each 
other, so unless otherwise instructed to, each container can access any 
hardware that the host system can. (well, almost) There is a simple 
permissions system to stop containers accessing hardware though (or 
allow). So - while it's not something I've tried myself (don't have 
additional PCI hardware in my LXC boxes) it ought to work, however the 
real experts are here: 
https://lists.sourceforge.net/lists/listinfo/lxc-users

Asterisk works just fine under LXC with internal timing, however LXC (at 
least the one I'm using) doesn't support real-time prioritys (the -p flag) 
that's not been an issue for me so-far though...

Gordon

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Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

2010-10-06 Thread Danny Dias
Thanks Steve,

I got the picture :) THANK!!!

But my doubt is about the cable, what cable should i use? i have a Sangoma
A108D in one machine (one machine with one card). What cable should i do?
how can i make it?

Best Regards!

2010/10/5 Steve Murphy m...@parsetree.com



 On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias ing.diasda...@gmail.comwrote:

 Hello my friend Ingmar,

 I would like to know the cable you used? how was the connection? i'm using
 this one:

 http://wiki.sangoma.com/Pinouts#A108 Loop Back

 Is this ok? what should i do my friend, my problems are understand the
 fisicall connection :(

 Best Regards!!!

 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl

  Hi DD,



 We usually use loopback cables and use the open source SIP test tool
 “SIPp” to initiate SIP calls that are sent from one group of 4 ports to
 another group of 4 ports.



 Met vriendelijke groet,

 Ingmar Steen

 Teleknowledge



 *Van:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias
 *Verzonden:* vrijdag 24 september 2010 11:05
 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Onderwerp:* [asterisk-users] How to test BIG traffic through
 DAHDI/WANPIPEinterfaces



 Hello Community,



 I need to test or simulate many calls through dahdi/wanpipe, i have a
 Sangoma A108D, and i need to test the stability of the
 card/drivers/firmwares with a test environment, do you think is possible?



 What should i do? using some loopback cable maybe?



 Thanks in advance



 DD


 I set up two machines with T1 interfaces, and connected the two with an
 appropriate t1 cable.
 One was acting as a network (master), the other as a subscriber (slave)
 (for timing). wrote two dialplans, one for each machine,
 that would answer an incoming call on one dahdi line, and call to the next
 numbered line on the other
 machine. The other machine was similarly outfit. I'd  define the extension
 for the first line on the t1,
 and call it with any phone you desire. That call will cascade into 23
 separate interlinked calls. If you are
 clever, the last call in should dial another real phone you have on-hand.

 You get the picture... right?   Phone A dials the exten to call the first
 exten on the other machine. The
 dialplan should use the first channel on the t1 to place a call to the
 first exten on the other machine.
 On the other machine, the incoming call on channel 1 is answered, and then
 a dial to the second extension
 on the first machine, over the 2nd t1 channel. The first machine answers,
 and uses the 3rd channel
 to call the other machine and so on till all channels are being used.
 The last exten answers and dials
 a phone (dahdi or SIP, no matter) that you pick up. Such a looped call
 should probably be awful, but
 it's going thru 23 t1 channels!

 If you have two t1 interaces in a single card (or two cards), then you do
 this on one machine.

 Another approach: set up equal numbers of FZO and FXS lines, and similarly
 loop s single call thru all the
 channels.This would require just one machine.

 Other approaches would involve running multiple threads to call an
 extension and then hang up and
 repeating this over and over again on all channels to ascertain the load
 placed just by call setup and tear-down.
 This kind of load is different than when all lines are just shoveling data
 back and forth.

 Watch your load averages, your %cpu, your swap, etc, as the tests are in
 full swing.

 murf






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 ParseTree Corp


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[asterisk-users] MYSQL ADDON INSTALLATION ERROR

2010-10-06 Thread Rizwan Hisham
Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years. I
hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
1.8.0-rc2 on centos 5.5 but getting the following errors.

app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory
app_mysql.c: In function ‘mysql_ds_destroy’:
app_mysql.c:135: warning: implicit declaration of function ‘mysql_close’
app_mysql.c:138: warning: implicit declaration of function
‘mysql_free_result’
app_mysql.c: In function ‘aMYSQL_connect’:
app_mysql.c:319: error: ‘MYSQL’ undeclared (first use in this function)
app_mysql.c:319: error: (Each undeclared identifier is reported only once
app_mysql.c:319: error: for each function it appears in.)
app_mysql.c:319: error: ‘mysql’ undeclared (first use in this function)

I think i have seen these errors before and did manage to get rid of them
but I cant remember how i did it and even dont remember the reason for these
errors. Looks like a header file for mysql addon is missing which is
actually missing (i have checked). How am I suppose to find it?

Plz help.

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Rizwan Qureshi
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Re: [asterisk-users] MYSQL ADDON INSTALLATION ERROR

2010-10-06 Thread Gareth Blades
Rizwan Hisham wrote:
 Hi All,
 Please refresh my memory. I am trying to install asterisk after 2 years. 
 I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 
 1.8.0-rc2 on centos 5.5 but getting the following errors.
 
 app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory
 app_mysql.c: In function ‘mysql_ds_destroy’:
 app_mysql.c:135: warning: implicit declaration of function ‘mysql_close’
 app_mysql.c:138: warning: implicit declaration of function 
 ‘mysql_free_result’
 app_mysql.c: In function ‘aMYSQL_connect’:
 app_mysql.c:319: error: ‘MYSQL’ undeclared (first use in this function)
 app_mysql.c:319: error: (Each undeclared identifier is reported only once
 app_mysql.c:319: error: for each function it appears in.)
 app_mysql.c:319: error: ‘mysql’ undeclared (first use in this function)
 
 I think i have seen these errors before and did manage to get rid of 
 them but I cant remember how i did it and even dont remember the reason 
 for these errors. Looks like a header file for mysql addon is missing 
 which is actually missing (i have checked). How am I suppose to find it?
 
 Plz help.
 
 -- 
 Best Regards
 Rizwan Qureshi
 
 

Make sure you have the mysql-devel package installed.

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Re: [asterisk-users] MYSQL ADDON INSTALLATION ERROR

2010-10-06 Thread Muhammad Nuzaihan Kamalluddin
Hi Ridwan,

You would need to install mysql-devel via yum.

Best Regards,
Muhammad Nuzaihan Kamal
Network Consultant
Mobile: +65 97473874

Asfa Systems Pte Ltd
91, Alps Avenue. #03-10. Singapore 498787

Tel:  +65 62538211
Fax: +65 62504814
www.asfasystems.com.sg

pub   4096R/36630777 2010-07-10
  Key fingerprint = 670A 4D60 0A2D 43A1 2FE0  DFDA D3A9 3F32 3663 0777
uid  Muhammad Nuzaihan Kamalluddin (Asfa Systems Pte. Ltd.) 
muham...@asfasystems.com
sub   4096R/97E5CBBD 2010-07-10



On 06-Oct-2010, at 6:35 PM, Rizwan Hisham wrote:

 Hi All,
 Please refresh my memory. I am trying to install asterisk after 2 years. I 
 hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 
 1.8.0-rc2 on centos 5.5 but getting the following errors.
 
 app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory
 app_mysql.c: In function ‘mysql_ds_destroy’:
 app_mysql.c:135: warning: implicit declaration of function ‘mysql_close’
 app_mysql.c:138: warning: implicit declaration of function ‘mysql_free_result’
 app_mysql.c: In function ‘aMYSQL_connect’:
 app_mysql.c:319: error: ‘MYSQL’ undeclared (first use in this function)
 app_mysql.c:319: error: (Each undeclared identifier is reported only once
 app_mysql.c:319: error: for each function it appears in.)
 app_mysql.c:319: error: ‘mysql’ undeclared (first use in this function)
 
 I think i have seen these errors before and did manage to get rid of them but 
 I cant remember how i did it and even dont remember the reason for these 
 errors. Looks like a header file for mysql addon is missing which is actually 
 missing (i have checked). How am I suppose to find it?
 
 Plz help.
 
 -- 
 Best Regards
 Rizwan Qureshi
 
 
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Re: [asterisk-users] MYSQL ADDON INSTALLATION ERROR

2010-10-06 Thread Steve Howes

On 6 Oct 2010, at 11:35, Rizwan Hisham wrote:

 Hi All,
 Please refresh my memory. I am trying to install asterisk after 2 years. I 
 hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 
 1.8.0-rc2 on centos 5.5 but getting the following errors.
 snip
 Plz help.

You need mysql-devel. You might also find that most things are case sensitive, 
maybe your malfunctioning caps-lock is causing problems? ;)

S
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Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?

2010-10-06 Thread Olivier
2010/10/5 Nikhil d.nik...@cem-solutions.net

  Hi
 We can run multiple instance of asterisk in same box with different IP
 and port. U need to install asterisk in different location eg: 1:
 /home/asterisk1/ 2 , : /home/asterisk2 ,and run both from that path ,
 listen ip and port should be different.

 command to run:

 $ /home/asterisk1/usr/sbin/asterisk -g for first asterisk
 $ /home/asterisk2/usr/sbin/asterisk -g for second asterisk


How do you then dealt with /etc/init.d/asterisk scripts ?
One per instance ?


 Thanks
 Nikhil


 On 10/05/2010 03:42 PM, bilal ghayyad wrote:
  Hi All;
 
  Did anyone try to implement (installation and configuration and running)
 for more than one asterisk instance (two or three instances), where each
 asterisk instance to work on a difference IP than the other where the server
 already has more than one IP address.
 
  We need to implement this situation because in case we need to do testing
 for any scenario of configuration, then other instances will not be
 effected.
 
  Is it possible without WMWare?
  Regards
  Bilal
 
 
 
 


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[asterisk-users] AMI getting related channels in Ringing state

2010-10-06 Thread Daniel Tryba
Issuing the AMI Status command results in a list of active channels. But
how to figure out which channels are related before the call is
answered? 2 channels below are somehow associated, but how can I be 100%
sure they are related in order to implement a redirect of the incoming
call to another phone (attended call pickup respecting
call/pickupgroups).

Uniqueid seems to be a timestamp, an IVR menu will render it useless for
this purpose. Same with the random part of the SIP channels
(177f/1780), this will not be sequential when multiple calls
occur with IVRs.

'SIP/922-1780' = array 
(
   'Privilege' = 'Call',
   'Channel' = 'SIP/922-1780',
   'CallerIDNum' = '922',
   'CallerIDName' = 'unknown',
   'Account' = '',
   'State' = 'Ringing',
   'Uniqueid' = '1286364290.8475',
)
'SIP/trunk-177f' = array 
(
   'Privilege' = 'Call',
   'Channel' = 'SIP/trunk-177f',
   'CallerIDNum' = 'trunk',
   'CallerIDName' = '0031234567890',
   'Accountcode' = '',
   'ChannelState' = '4',
   'ChannelStateDesc' = 'Ring',
   'Context' = 'macro-dial-one',
   'Extension' = 's',
   'Priority' = '37',
   'Seconds' = '3',
   'Uniqueid' = '1286364290.8474',
)

-- 

   Daniel Tryba

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Re: [asterisk-users] MYSQL ADDON INSTALLATION ERROR

2010-10-06 Thread Rizwan Hisham
Thank you all. It is now installed.

On Wed, Oct 6, 2010 at 5:04 PM, Steve Howes steve-li...@geekinter.netwrote:


 On 6 Oct 2010, at 11:35, Rizwan Hisham wrote:

  Hi All,
  Please refresh my memory. I am trying to install asterisk after 2 years.
 I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
 1.8.0-rc2 on centos 5.5 but getting the following errors.
  snip
  Plz help.

 You need mysql-devel. You might also find that most things are case
 sensitive, maybe your malfunctioning caps-lock is causing problems? ;)

 S
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Rizwan Qureshi
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Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

2010-10-06 Thread Ryan Wagoner
The Loop Back Plug on the link you provided is correct. You take a
few inches of CAT5 and remove the outer jacket. Loop the wires into
the RJ-45 connector like the diagram shows and then crimp.

Ryan

On Tue, Oct 5, 2010 at 3:02 PM, Danny Dias ing.diasda...@gmail.com wrote:
 Hello my friend Ingmar,
 I would like to know the cable you used? how was the connection? i'm using
 this one:
 http://wiki.sangoma.com/Pinouts#A108 Loop Back
 Is this ok? what should i do my friend, my problems are understand the
 fisicall connection :(
 Best Regards!!!

 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl

 Hi DD,



 We usually use loopback cables and use the open source SIP test tool
 “SIPp” to initiate SIP calls that are sent from one group of 4 ports to
 another group of 4 ports.



 Met vriendelijke groet,

 Ingmar Steen

 Teleknowledge



 Van: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias
 Verzonden: vrijdag 24 september 2010 11:05
 Aan: Asterisk Users Mailing List - Non-Commercial Discussion
 Onderwerp: [asterisk-users] How to test BIG traffic through
 DAHDI/WANPIPEinterfaces



 Hello Community,



 I need to test or simulate many calls through dahdi/wanpipe, i have a
 Sangoma A108D, and i need to test the stability of the
 card/drivers/firmwares with a test environment, do you think is possible?



 What should i do? using some loopback cable maybe?



 Thanks in advance



 DD

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[asterisk-users] Difference

2010-10-06 Thread Rizwan Hisham
Hi All,
Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk
versions.

Thanks

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[asterisk-users] Page minimum number of extensions

2010-10-06 Thread Matteo Fortini
Hi,
if I Page more than one extension, then the MeetMe conference stays up 
even if all the called extensions aren't available or are hung up.
Is there a way of keeping track of how many extensions are attached to 
the conference, and require a number or a particular extension to be 
present?

Thank you

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Re: [asterisk-users] Cisco SIP 8.5 and 9.0 Issues

2010-10-06 Thread Gerard
Hi James,
I have a smartNet as well, I contacted cisco for a login, so when I go 
to the support section for 7900 series phones, then hit Download 
Software, it lists this for the 7962G :
Expand all | Close all
Latest Releases
9.0(3)
All Releases
SIP v.9
9.0(3)
9.0(2)SR2
9.0(2)SR1
SIP v.8
8.5(4)
8.5(3)SR1
8.5(3)
8.5(2)SR1
8.5(2)
8.4(4)
8.4(3)
8.4(2)
8.4(1)_SR2
8.4(1)
8.3(5)
8.3(4)_SR1
8.3(3)_SR2
8.3(3)
8.3(2)_SR1
8.3(2)

kind of an ugly cutpaste, but you get the idea..

now I also have a 7960 phone, and I have the 8.12 firmware loaded on it, 
and it works 99.9% flawless, it just 'forgets' its custom ringtone on 
occasion.
but as you can see, no 8.12 is available for the 7962G..

-Gerard


On 10/05/10 16:55, James Miller wrote:
 I know this doesn't answer your question directly, but Where are you
 getting the Sip 9.0 software? It is not available on Cisco's website.

 I have Sip 8.9 on my phone and I have noticed that after about 45 mins
 on a call it will hang up and drop the desktop connection that runs
 through the phone.

 I am hoping that upgrading to 8.12 will fix the issue, or I just wasted
 money on a SMARTnet contract.

 Regards,
 James


 I see blindness, not as a disability, but more of an ability.  And
 Sight actually, more of a disability because some people with sight tend
 to judge others by what they see on the outside, whereas I don't see
 that. I just see that which is in a person.  Patrick Henry Hughes,
 Louisville Kentucky,2008


 On Tue, Oct 5, 2010 at 17:08, Gerard gsara...@rarcoa.com
 mailto:gsara...@rarcoa.com wrote:

 Hi list,
 I was wondering if anyone had any solution to either one of two issues
 I'm having:
 I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware,
 it works very well for the most part, but after less then a week of
 heavy usage, eventually the phone gets into a state where it won't
 accept or let you place any more calls, the screen flashes no free
 lines available or something along those lines. (power cycle fixes
 this).
 So my preferred solution would be to upgrade to the v9.0(3) firmware,
 but when that's loaded, the phone won't register with Asterisk anymore,
 does anyone know if I need to adjust my .cnf.xml file, or is it a bug of
 some sort?
 Thanks for any input,
 --
 Gerard Saraber
 Network Admin.
 Rarcoa, Inc

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Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

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Re: [asterisk-users] Cisco SIP 8.5 and 9.0 Issues

2010-10-06 Thread JR Richardson
 Hi list,
 I was wondering if anyone had any solution to either one of two issues
 I'm having:
 I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware,
 it works very well for the most part, but after less then a week of
 heavy usage, eventually the phone gets into a state where it won't
 accept or let you place any more calls, the screen flashes no free
 lines available or something along those lines. (power cycle fixes this).
 So my preferred solution would be to upgrade to the v9.0(3) firmware,
 but when that's loaded, the phone won't register with Asterisk anymore,
 does anyone know if I need to adjust my .cnf.xml file, or is it a bug of
 some sort?
 Thanks for any input,
 --
 Gerard Saraber
 Network Admin.
 Rarcoa, Inc

Use Polycom, but if you really must use cisco phones, downgrade to 7.5.
I've got a lot of 79xx phones out there and 7.5 is the last stable release
as far as I'm concerned.  It just seems to work, no periodic reboots needed,
or any other quirkiness like with the newer firmware's.  The feature set is
not robust, but it is reliable with Asterisk.  Keep in mind, Cisco has no
incentive to make their SIP firmware work with any other platform other than
there servers so I don't really expect it to work properly if at all with
Asterisk.  7.5 is the only firmware version that I deploy on a few hundred
units and works fine.

Good luck.

JR


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Re: [asterisk-users] Cisco SIP 8.5 and 9.0 Issues

2010-10-06 Thread Gerard
I think you're right JR,
that has been my experience as well, I've run just great with sip 
7.something firmwares for years.. unfortunately those don't support the 
wireless headsets that the people I support want.
My boss really likes the cisco phones, the hardware feels really solid, 
I wouldn't mind testing out a polycom phone though.

The chan-sccp guys are really awesome, it's just not quite ready for my 
office at the moment, it's getting there though, maybe that's a better 
bet. Especially if I can't get SIP sorted out.

-Gerard

On 10/06/10 07:56, JR Richardson wrote:
 Hi list,
 I was wondering if anyone had any solution to either one of two issues
 I'm having:
 I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware,
 it works very well for the most part, but after less then a week of
 heavy usage, eventually the phone gets into a state where it won't
 accept or let you place any more calls, the screen flashes no free
 lines available or something along those lines. (power cycle fixes this).
 So my preferred solution would be to upgrade to the v9.0(3) firmware,
 but when that's loaded, the phone won't register with Asterisk anymore,
 does anyone know if I need to adjust my .cnf.xml file, or is it a bug of
 some sort?
 Thanks for any input,
 --
 Gerard Saraber
 Network Admin.
 Rarcoa, Inc

 Use Polycom, but if you really must use cisco phones, downgrade to 7.5.
 I've got a lot of 79xx phones out there and 7.5 is the last stable release
 as far as I'm concerned.  It just seems to work, no periodic reboots needed,
 or any other quirkiness like with the newer firmware's.  The feature set is
 not robust, but it is reliable with Asterisk.  Keep in mind, Cisco has no
 incentive to make their SIP firmware work with any other platform other than
 there servers so I don't really expect it to work properly if at all with
 Asterisk.  7.5 is the only firmware version that I deploy on a few hundred
 units and works fine.

 Good luck.

 JR




-- 
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Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)

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[asterisk-users] Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold

2010-10-06 Thread Karsten Wemheuer
Hi,

while testing current release candidate 1.8.0-rc2 I stumbled on a weird
behavior. I did not find any hints in the archives or at the bug
tracker.

Two SIP-Clients are connected (both on the local net, no NAT). The RTP
stream flows directly between the phones. If I set phone A on hold, the
music on hold is played. On the CLI I see the following message running:
WARNING[2470]: res_rtp_asterisk.c:1939 bridge_p2p_rtp_write: RTP
Transmission error of packet to (null): Invalid argument

The message is running until the phones are connected again. In the
meantime the CLI is nearly unusable. This does not happen, if I
configure asterisk to stay in the media path.

Is this a new bug or do I something wrong?

File sip.conf looks like this:

[general]
bindaddr = 0.0.0.0
disallow = all
allow = alaw
allow = ulaw
language = de
allowguest = no
fromdomain = 192.168.10.70
tos_sip = 96
tos_audio = 184

[katrin]
type = friend
host = dynamic
callerid = Katrin Wemheuer 200
context = Standard
mailbox = 200

[max]
type = friend
host = dynamic
callerid = Max Müller 245
context = Standard
mailbox = 245

Thanks,

Karsten


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Re: [asterisk-users] AMI getting related channels in Ringing state

2010-10-06 Thread Daniel Tryba
On Wed, Oct 06, 2010 at 01:56:55PM +0200, Daniel Tryba wrote:
 Issuing the AMI Status command results in a list of active channels. But
 how to figure out which channels are related before the call is
 answered?

CoreShowChannels gives a little bit of extra data in the originator
channel:

Application: Dial
ApplicationData: SIP/922

This makes finding the actual ringing SIP device fun when the incoming
channel dials a Local/9...@default device. And doesn't give any usefull
hint when the incoming channel doesn't use Dial but e.g. Queue.

-- 

   Daniel Tryba

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Re: [asterisk-users] Difference

2010-10-06 Thread Danny Nicholas
  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Wednesday, October 06, 2010 7:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Difference

 

Hi All,
Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk
versions.

Thanks

-- 
Best Regards
Rizwan Qureshi



In a nutshell, 1.4 is the oldest and most stable, 1.6 is the current and 1.8
is the beta version of Asterisk.  This is a gross over-simplification, but
if you know nothing, 1.4 is going to give you the fewest headaches and if
you have to have the latest 1.6 or 1.8 is the way to go.  The ChangeLogs
on Asterisk.org will give you a detailed difference.

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Re: [asterisk-users] Cisco SIP 8.5 and 9.0 Issues

2010-10-06 Thread Peder
Use Polycom, but if you really must use cisco phones, downgrade to 7.5.
I've got a lot of 79xx phones out there and 7.5 is the last stable release
as far as I'm concerned.  It just seems to work, no periodic reboots
needed,
or any other quirkiness like with the newer firmware's.  The feature set is
not robust, but it is reliable with Asterisk.  Keep in mind, Cisco has no
incentive to make their SIP firmware work with any other platform other
than
there servers so I don't really expect it to work properly if at all with
Asterisk.  7.5 is the only firmware version that I deploy on a few hundred
units and works fine.

I second that.  We have been using 7.5 with a couple hundred phones for
years with no issues.


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Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware

2010-10-06 Thread Andrew Latham
It is describing the method of copying the German localization over
the United_States_English so the phones use German



~
Andrew lathama Latham
lath...@gmail.com

* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux



On Wed, Oct 6, 2010 at 3:10 AM, Olivier oza_4...@yahoo.fr wrote:


 2010/10/5 Andrew Latham lath...@gmail.com

 http://www.ip-phone-forum.de/showthread.php?t=188877


 For those not fluent in german, what is this thread telling ?


 ~
 Andrew lathama Latham
 lath...@gmail.com

 * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
 * Learn more about Linux http://en.wikipedia.org/wiki/Linux
 * Learn more about Tux http://en.wikipedia.org/wiki/Tux



 On Tue, Jul 3, 2007 at 1:24 PM, Olivier oza-4...@myamail.com wrote:
  Any reply ?
 
  2007/7/1, Olivier oza-4...@myamail.com:
 
  Thanks everybody for your input.
 
  Let me summarize localization process :
 
  1. Buring boot, phones download from TFTP server an xml or older .cfg
  file
  in which a localization parameter is set.
  2. When this parameter is read, phones will then ask CCM or Asterisk or
  another server to send localized button templates using SIP messages.
  3. When SIP messages are received, phones will then localize buttons
  and
  menus.
  4. Whenever an incoming or outgoing call is processed, phones will no
  further ask any localization data from CCM or Asterisk or anything as
  localization was fully done during initialization process.
 
  A. Is this correct ?
  Specifically, shall I understand that if present initialization process
  is
  extended to include localization, there will be no need to change
  Asterisk
  SIP channel to fully support localized Cisco phones ?
 
  B. What are the SIP messages that trigger and reply to localization
  demands ?
  Is it possible to tailor dialplan so that these Cisco specific messages
  are treated without modifying chan-sip ?
 
  Cheers
 
 
 
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[asterisk-users] using better quality wav or mp3 in Asterisk 1.2.x

2010-10-06 Thread Zarko Zivanovic
Hello,

I would need a little help about using 16 bit wav or mp3 files for moh on
asterisk 1.2.x

When i try to use these files as moh, the caller gets disconnected.

 

Please advise.

Regards,

Z. Zivanovic

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Re: [asterisk-users] How to test BIG traffic throughDAHDI/WANPIPEinterfaces

2010-10-06 Thread Ingmar Steen
Hi Danny,

 

We're using A104 cards which reduces the complexity of the necessary
cable (it's a regular T1/E1 cross cable).

 

In case of a A108, the cable that would come closest to what we're doing
is combining the A108 Straight Thru Y Cable and the A108 Cross-Over Y
Cable (or getting both and connecting them with straight thru F-F
connectors).

 

Met vriendelijke groet,

Ingmar Steen

Teleknowledge

 

Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias
Verzonden: woensdag 6 oktober 2010 11:12
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] How to test BIG traffic
throughDAHDI/WANPIPEinterfaces

 

Thanks Steve,

 

I got the picture :) THANK!!!

 

But my doubt is about the cable, what cable should i use? i have a
Sangoma A108D in one machine (one machine with one card). What cable
should i do? how can i make it?

 

Best Regards!

 

2010/10/5 Steve Murphy m...@parsetree.com

 

On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias ing.diasda...@gmail.com
wrote:

Hello my friend Ingmar,

 

I would like to know the cable you used? how was the connection? i'm
using this one:

 

http://wiki.sangoma.com/Pinouts#A108 Loop Back

 

Is this ok? what should i do my friend, my problems are understand the
fisicall connection :(

 

Best Regards!!!

2010/9/24 Ingmar Steen i.st...@teleknowledge.nl

Hi DD,

 

We usually use loopback cables and use the open source SIP test tool
SIPp to initiate SIP calls that are sent from one group of 4 ports to
another group of 4 ports.

 

Met vriendelijke groet,

Ingmar Steen

Teleknowledge

 

Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias
Verzonden: vrijdag 24 september 2010 11:05
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [asterisk-users] How to test BIG traffic through
DAHDI/WANPIPEinterfaces

 

Hello Community,

 

I need to test or simulate many calls through dahdi/wanpipe, i have a
Sangoma A108D, and i need to test the stability of the
card/drivers/firmwares with a test environment, do you think is
possible?

 

What should i do? using some loopback cable maybe?

 

Thanks in advance

 

DD  


I set up two machines with T1 interfaces, and connected the two with an
appropriate t1 cable.
One was acting as a network (master), the other as a subscriber (slave)
(for timing). wrote two dialplans, one for each machine,
that would answer an incoming call on one dahdi line, and call to the
next numbered line on the other
machine. The other machine was similarly outfit. I'd  define the
extension for the first line on the t1,
and call it with any phone you desire. That call will cascade into 23
separate interlinked calls. If you are
clever, the last call in should dial another real phone you have
on-hand.

You get the picture... right?   Phone A dials the exten to call the
first exten on the other machine. The
dialplan should use the first channel on the t1 to place a call to the
first exten on the other machine.
On the other machine, the incoming call on channel 1 is answered, and
then a dial to the second extension
on the first machine, over the 2nd t1 channel. The first machine
answers, and uses the 3rd channel
to call the other machine and so on till all channels are being
used. The last exten answers and dials
a phone (dahdi or SIP, no matter) that you pick up. Such a looped call
should probably be awful, but
it's going thru 23 t1 channels!

If you have two t1 interaces in a single card (or two cards), then you
do this on one machine.

Another approach: set up equal numbers of FZO and FXS lines, and
similarly loop s single call thru all the
channels.This would require just one machine.

Other approaches would involve running multiple threads to call an
extension and then hang up and
repeating this over and over again on all channels to ascertain the load
placed just by call setup and tear-down.
This kind of load is different than when all lines are just shoveling
data back and forth.

Watch your load averages, your %cpu, your swap, etc, as the tests are in
full swing.

murf



 


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Re: [asterisk-users] Repeated: Got SIP response 489 Bad eventback from

2010-10-06 Thread Gopalakrishnan A.N
Hi James,

I too facing the same issue whereas in the inbound call I am able to receive
the call, when I pickup the receiver it hangsup. I am getting the NOTIFY
option.. the log as follows,

-- SIP read from 98.158.181.173:5060:
NOTIFY sip:pbxfami...@10.0.8.84:5060 SIP/2.0
Via: SIP/2.0/UDP 98.158.181.173:5060;branch=z9hG4bK47ff44c5;rport
From: Unknown sip:unkn...@98.158.181.173 sip%3aunkn...@98.158.181.173
;tag=as24f09d54
To: sip:p...@10.0.8.84:5060
Contact: sip:unkn...@98.158.181.173 sip%3aunkn...@98.158.181.173
Call-ID: 03d488a828e9bee61ca72fc16f378...@98.158.181.173
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 89

Messages-Waiting: no
Message-Account: sip:*...@98.158.181.173
Voice-Message: 0/0 (0/0)

Oct  6 08:57:43 VERBOSE[31038] logger.c: --- (12 headers 3 lines)Oct  6
08:57:43 VERBOSE[31038] logger.c: --- (12 headers 3 lines)---
Oct  6 08:57:43 VERBOSE[31038] logger.c: Trasmitting Response: 489 Bad event
Oct  6 08:57:43 VERBOSE[31038] logger.c: HERE chan_sip.c ast_sip_ouraddrfor
1365
Oct  6 08:57:43 VERBOSE[31038] logger.c: Transmitting (no NAT) to
98.158.181.173:5060:
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 98.158.181.173:5060
;branch=z9hG4bK47ff44c5;rport;received=98.158.181.173
From: Unknown sip:unkn...@98.158.181.173 sip%3aunkn...@98.158.181.173
;tag=as24f09d54
To: sip:pbxfami...@10.0.8.84:5060;tag=as3cdb539f
Call-ID: 03d488a828e9bee61ca72fc16f378...@98.158.181.173
CSeq: 102 NOTIFY
User-Agent: CEM PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0

My Setup: I have created one extension in elastix and the extension is
configured as VoIP trunk in Asterisk.




On Sun, Apr 11, 2010 at 1:10 PM, Adrian Marsh adrian.ma...@ubiquisys.comwrote:

 Hi James,

 Thanks for the help.  3.10 registers into my SIP server just as a normal
 SIP client.
 Yes, qualify=yes.   I just tried setting that to no on my end, and I still
 get the message. I'll try turning it off on 3.10 too tomorrow and capture
 some trace too

 Adrian

  Hi All,
 
 
 
  I've two asterisk servers on the same LAN, both 1.4, and I keep getting
 Got
  SIP response 489 Bad event back from 192.168.3.10
 
  No idea whats causing it. The only references I can find mentions NATing
  issues, but these are on the same LAN so NAT shouldn't be an issue.
 
  3.10 does authenticate into the server logging the error.  The error
 appears
  in the log every 1m20s (ish)

 Is 3.10 on a SIP trunk to the other asterisk box?
 Is qualify=yes on this SIP trunk?
 I think you'll find that if you run an ngrep/tcpdump on port 5060 on
 the box receiving the error it will send out an OPTIONS or NOTIFY (I
 can't remember which) and then you'll see the 489 Bad Event.
 Grab a trace of the SIP traffic and post it, its the only way to know
 for sure though.

 -- James

 
 
 
  Any ideas?
 
 
 
  Thanks,
 
 
 
  Adrian
 
 
 
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http://lists.digium.com/mailman/listinfo/asterisk-users
 

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[asterisk-users] 2 way intercom recommendation for restaurant kitchens

2010-10-06 Thread Andy Graybeal
Greetings,
I need a 2 way intercom for separate kitchens to communicate without 
having to walk back and forth.

The speaker has to be loud but clear, not distorted.  Sometimes the 
kitchens can be noisy.

It needs to be easy to use.

It needs to be easy to clean.

It would be nice if it used POE.

Eventually I would like the kitchens to be able to dial different parts 
of the restaurant when I get the whole place switched to VOIP, but for 
now I need something only in the two kitchens.

I like the idea of a regular phone with a kick'n speakerphone, but I'm 
open to alternatives.  I say 'regular phone' with unease, but I mean 
something with a normal dialpad, extra buttons for different functions, 
handset and speakerphone.

I've been considering cisco and polycom.  Specifically I've been 
thinking about the Cisco 7940g or something like it.  Also I've been 
considering the Cisco 7920 in a holster w/ wired headset.

I'm welcome to any recommendations.

thank you,
-Andy

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Re: [asterisk-users] 2 way intercom recommendation for restaurantkitchens

2010-10-06 Thread Danny Nicholas
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 10:07 AM
To: asterisk-users@lists.digium.com
Cc: t...@casanueva.com
Subject: [asterisk-users] 2 way intercom recommendation for
restaurantkitchens

Greetings,
I need a 2 way intercom for separate kitchens to communicate without 
having to walk back and forth.

The speaker has to be loud but clear, not distorted.  Sometimes the 
kitchens can be noisy.

It needs to be easy to use.

It needs to be easy to clean.

It would be nice if it used POE.

Eventually I would like the kitchens to be able to dial different parts 
of the restaurant when I get the whole place switched to VOIP, but for 
now I need something only in the two kitchens.

I like the idea of a regular phone with a kick'n speakerphone, but I'm 
open to alternatives.  I say 'regular phone' with unease, but I mean 
something with a normal dialpad, extra buttons for different functions, 
handset and speakerphone.

I've been considering cisco and polycom.  Specifically I've been 
thinking about the Cisco 7940g or something like it.  Also I've been 
considering the Cisco 7920 in a holster w/ wired headset.

I'm welcome to any recommendations.

thank you,
-Andy

Polycom 501's are pretty good and relatively inexpensive.


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Re: [asterisk-users] Difference

2010-10-06 Thread Rizwan Hisham
Is there any major architectural difference between 1.4 and 1.8?

On Wed, Oct 6, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote:

   --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
 *Sent:* Wednesday, October 06, 2010 7:15 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Difference



 Hi All,
 Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk
 versions.

 Thanks

 --
 Best Regards
 Rizwan Qureshi

  In a nutshell, 1.4 is the oldest and most stable, 1.6 is the current and
 1.8 is the beta version of Asterisk.  This is a gross over-simplification,
 but if you “know nothing”, 1.4 is going to give you the fewest headaches and
 if you “have to have the latest” 1.6 or 1.8 is the way to go.  The
 ChangeLogs on Asterisk.org will give you a detailed difference.

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   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Best Regards
Rizwan Qureshi
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Re: [asterisk-users] Difference

2010-10-06 Thread Danny Nicholas
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Wednesday, October 06, 2010 7:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Difference

 

Hi All,
Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk
versions.

Thanks

-- 
Best Regards
Rizwan Qureshi

In a nutshell, 1.4 is the oldest and most stable, 1.6 is the current and 1.8
is the beta version of Asterisk.  This is a gross over-simplification, but
if you know nothing, 1.4 is going to give you the fewest headaches and if
you have to have the latest 1.6 or 1.8 is the way to go.  The ChangeLogs
on Asterisk.org will give you a detailed difference.




  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Wednesday, October 06, 2010 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Difference

 

Is there any major architectural difference between 1.4 and 1.8?

The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from ,
to |) and the AGI structure is enhanced.  If you don't use AGI's, a
qualified not really.

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Re: [asterisk-users] Difference

2010-10-06 Thread Steve Edwards
On Wed, 6 Oct 2010, Rizwan Hisham wrote:

 Is there any major architectural difference between 1.4 and 1.8?

Nope. The developer's just got tired of typing .4

Of course, the joke's on them -- 1.8 is only .4 better than 1.4.

-- 
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] 2 way intercom recommendation for restaurantkitchens

2010-10-06 Thread Andy Graybeal
 Polycom 501's are pretty good and relatively inexpensive.



Danny,
Should I be worried that the Polycom 501 has been discontinued?  What 
does this even imply... that they won't be putting out any BIOS updates 
(if there even is a BIOS on phones...)

Sounds like they'd be cheap to get on ebay though :)

-Andy

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Re: [asterisk-users] Difference

2010-10-06 Thread Zeeshan Zakaria
For a production environment, 1.4 is the most stable, and it has everything
one needs to setup a telecom platform. As per my understanding 1.6 never got
the same recognition for stability as 1.4, plus it doesn't have any
significant advantages over 1.4. The newer version 1.8 series might be my
next jump once it'll be out of beta, but at this time it should not be used
in a production environment. Many of us still use 1.4 in production and if
you are just starting, this'll be your best choice.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-10-06 11:54 AM, Danny Nicholas da...@debsinc.com wrote:

 From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Be...

*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
*Sent:* Wednesday, October 06, 2010 10:44 AM


To: Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Difference





Is there any major architectural difference between 1.4 and 1.8?

The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from ,
to |) and the AGI structure is enhanced.  If you don’t use AGI’s, a
qualified “not really”.

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  http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] 2 way intercom recommendationfor restaurantkitchens

2010-10-06 Thread Danny Nicholas
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 way intercom recommendationfor
restaurantkitchens

 Polycom 501's are pretty good and relatively inexpensive.



Danny,
Should I be worried that the Polycom 501 has been discontinued?  What 
does this even imply... that they won't be putting out any BIOS updates 
(if there even is a BIOS on phones...)

Sounds like they'd be cheap to get on ebay though :)

-Andy

It is already a relative PITA to get BIOS updates - that being said, when
you are able to get them, there are plenty and Polycom is pretty good about
updating BIOS for discontinued phones.


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Re: [asterisk-users] Difference

2010-10-06 Thread Rizwan Hisham
Back in the days i heard that they have changed the architecture in 1.6 and
its a lot better than 1.4 (6 times better call handling and robust
architecture, someone told me). If they have decided to take the 1.6
architecture to the next level in the new 1.8 version then its a good thing.

On Wed, Oct 6, 2010 at 9:58 PM, Steve Edwards asterisk@sedwards.comwrote:

 On Wed, 6 Oct 2010, Rizwan Hisham wrote:

  Is there any major architectural difference between 1.4 and 1.8?

 Nope. The developer's just got tired of typing .4

 Of course, the joke's on them -- 1.8 is only .4 better than 1.4.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

 --
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-- 
Best Regards
Rizwan Qureshi
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Re: [asterisk-users] Difference

2010-10-06 Thread Zeeshan Zakaria
Here is a presentation from Kevin P. Fleming, Director of Software
Technologies at Digium. Information might be old by now still gives a good
overview of what is new in 1.6:

http://www.asterisk-tag.org/2008/slides/Kevin-Fleming-Asterisk-Tag-2008.pdf

Summary of his presentation is as follows:

– Asterisk 1.6 contains much new functionality, although nothing
revolutionary
– Asterisk 1.6's core has been improved in many ways that will reduce the
performance impact of new features being added and also the likelihood
of difficult to find locking and data structure bugs
– Future releases of Asterisk 1.6 (1.6.1, 1.6.2, etc.) will get new
functionality as well, in a controlled fashion
– Asterisk 1.6.0 is not recommended for production usage yet, but we would
very much like users to try it, report problems and help test the product in
more scenarios than the development can test themselves

--

Zeeshan

On Wed, Oct 6, 2010 at 12:12 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:

 Back in the days i heard that they have changed the architecture in 1.6 and
 its a lot better than 1.4 (6 times better call handling and robust
 architecture, someone told me). If they have decided to take the 1.6
 architecture to the next level in the new 1.8 version then its a good thing.


 On Wed, Oct 6, 2010 at 9:58 PM, Steve Edwards 
 asterisk@sedwards.comwrote:

 On Wed, 6 Oct 2010, Rizwan Hisham wrote:

  Is there any major architectural difference between 1.4 and 1.8?

 Nope. The developer's just got tired of typing .4

 Of course, the joke's on them -- 1.8 is only .4 better than 1.4.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

 --
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   http://lists.digium.com/mailman/listinfo/asterisk-users




 --
 Best Regards
 Rizwan Qureshi



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-- 
Zeeshan A Zakaria
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Re: [asterisk-users] Difference

2010-10-06 Thread Miguel Molina
I find 1.6.2.13 version is stable for trunk call routing, and it should 
be too for basic call center use. The asterisk team has made some 
architectural improvements (moving to astobj2 a lot of internal 
structures, and much more you may not see from a user perspective) but 
given the several environment and different use cases, fear to upgrade 
or proven 1.4 stability for the job, the people usually don't upgrade or 
make it slowly with a lot of previous tests before making the jump.


If you use FAX, I recommend you 1.6.2 or later. The app_fax module is 
far better than the ast-agx-addons for 1.4.


The good old (now unsupported) 1.2 works for many people, ask Steve.

So it's up to you.

Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center

El 06/10/10 11:04, Zeeshan Zakaria escribió:


For a production environment, 1.4 is the most stable, and it has 
everything one needs to setup a telecom platform. As per my 
understanding 1.6 never got the same recognition for stability as 1.4, 
plus it doesn't have any significant advantages over 1.4. The newer 
version 1.8 series might be my next jump once it'll be out of beta, 
but at this time it should not be used in a production environment. 
Many of us still use 1.4 in production and if you are just starting, 
this'll be your best choice.


Zeeshan A Zakaria

--
www.ilovetovoip.com http://www.ilovetovoip.com

On 2010-10-06 11:54 AM, Danny Nicholas da...@debsinc.com 
mailto:da...@debsinc.com wrote:


From: asterisk-users-boun...@lists.digium.com 
mailto:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com 
mailto:asterisk-users-boun...@lists.digium.com] On Be...


*From:* asterisk-users-boun...@lists.digium.com 
mailto:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com 
mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Rizwan Hisham

*Sent:* Wednesday, October 06, 2010 10:44 AM


To: Asterisk Users Mailing List - Non-Commercial Discussion

*Subject:* Re: [asterisk-users] Difference





Is there any major architectural difference between 1.4 and 1.8?

The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes 
from , to |) and the AGI structure is enhanced.  If you don’t use 
AGI’s, a qualified “not really”.



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Re: [asterisk-users] 2 way intercom recommendationfor restaurantkitchens

2010-10-06 Thread Steve Edwards
On Wed, 6 Oct 2010, Danny Nicholas wrote:

 Polycom 501's are pretty good and relatively inexpensive.

 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy 
 Graybeal

 Should I be worried that the Polycom 501 has been discontinued?  What 
 does this even imply... that they won't be putting out any BIOS updates 
 (if there even is a BIOS on phones...)

 Sounds like they'd be cheap to get on ebay though :)

On Wed, 6 Oct 2010, Danny Nicholas wrote:

 It is already a relative PITA to get BIOS updates - that being said, 
 when you are able to get them, there are plenty and Polycom is pretty 
 good about updating BIOS for discontinued phones.

It has been my experience that it is easier to get Polycom firmware 
updates (just download off their web site) than just about anybody 
coughcisco suckscough else.

Polycom even has PDFs on how to set up TFTP, FTP, or HTTP servers and 
guides on configuring their gear to work with Asterisk.

In a restaurant environment, I'd be looking for some sort of faceplate 
overlay to make cleanup easier.

-- 
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] using better quality wav or mp3 in Asterisk 1.2.x

2010-10-06 Thread Kyle Kienapfel
On Wed, Oct 6, 2010 at 7:03 AM, Zarko Zivanovic outlaw...@gmail.com wrote:

  Hello,

 I would need a little help about using 16 bit wav or mp3 files for moh on
 asterisk 1.2.x

 When i try to use these files as moh, the caller gets disconnected.



 Please advise.

 Regards,

 Z. Zivanovic


 __ Information from ESET NOD32 Antivirus, version of virus
 signature database 5509 (20101006) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com

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What codec(s) are your calls using? You would be better off converting them
before hand.

Also, what is sample rate on your wav/mp3 files? Asterisk is picky about
wanting 8000hz sample rate. Newer asterisk may support 16khz sample rate as
well. You are most likely dealing with 44.1khz material.
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Re: [asterisk-users] Asterisk sharing a line with POTS handsets: how to interoperate cleanly?

2010-10-06 Thread Kyle Kienapfel
On Tue, Oct 5, 2010 at 1:40 PM, Roger Burton West ro...@firedrake.orgwrote:

 I now have an OpenVox A400P and it is working well. Thanks to Ade
 Vickers for the recommendation, which I second.

 However, I need to make a slow transition between a conventional
 multiple-extension setup and a full VoIP network on these premises. So
 at the moment the Asterisk box shares the PSTN connection with several
 conventional analogue handsets.

 The desired result for an incoming call is that the Asterisk server will
 wait N seconds before answering (which I can arrange easily enough), and
 if the call has been answered on one of the handsets by that time the
 Asterisk server should ignore it completely. Otherwise it should start
 checking CLID, prompting for extensions, and other good stuff, which
 again I know how to do.

 What is a good approach to making sure the Asterisk server doesn't pick
 up a call that has been answered elsewhere? (Ideally in pure dialplan,
 but a perl AGI would also do.)

 R

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I'd start with noting down what happens by default. Then you'll know exactly
what behavior to target.

I was worried about similar issues with my SPA3102, but if I pick up a call
from a regular phone before it picks up, it just assumes the person on the
other end gave up. Also it wont pick up the line if it is in use.
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Re: [asterisk-users] 2 way intercom recommendationfor restaurantkitchens

2010-10-06 Thread Andy Graybeal
 It is already a relative PITA to get BIOS updates - that being said,
 when you are able to get them, there are plenty and Polycom is pretty
 good about updating BIOS for discontinued phones.

 It has been my experience that it is easier to get Polycom firmware
 updates (just download off their web site) than just about anybody
 coughcisco suckscough  else.

 Polycom even has PDFs on how to set up TFTP, FTP, or HTTP servers and
 guides on configuring their gear to work with Asterisk.

 In a restaurant environment, I'd be looking for some sort of faceplate
 overlay to make cleanup easier.


Okay great, Polycom it is then.

The 501, does it have a good speakerphone?
Something loud and clear, no distortion?
Something that would be good with quite a bit of background noise.
Do I need to lift the handset to hear the speakerphone better?  From 
pictures online it looks like the speaker is under the handset which 
doesn't look very intuitive.

I'm hoping the cooks only have to hit one button and be able to reach 
the other kitchen, no fumbling with the handset.  I guess I can remove 
the handset.

Faceplates.. interesting, a quick search on 'polycom 501 faceplate' or 
'polycom 501 stainless steel faceplate' in google doesn't come back very 
enthusiastic.  Is there such a thing?

Thank you so far for the feedback.  It's made me feel more confident and 
excited.

-Andy

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[asterisk-users] AMI connection limit

2010-10-06 Thread Jerry Geis
Is there a limit to the AMI connections?
I have upto 3 process that might connect and originate calls,
2 of those processes are very infrequent - however the main one
can issue a number of calls one right after the other.

I am seeing a message when I am logging failed manager connection 
sometimes.

Are there any limits I could be running into? It does start responding 
again with no changes.
I am running 1.4.X

Thanks,

Jerry

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Re: [asterisk-users] 2 way intercom recommendationforrestaurantkitchens

2010-10-06 Thread Danny Nicholas
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 way intercom
recommendationforrestaurantkitchens

 It is already a relative PITA to get BIOS updates - that being said,
 when you are able to get them, there are plenty and Polycom is pretty
 good about updating BIOS for discontinued phones.

 It has been my experience that it is easier to get Polycom firmware
 updates (just download off their web site) than just about anybody
 coughcisco suckscough  else.

 Polycom even has PDFs on how to set up TFTP, FTP, or HTTP servers and
 guides on configuring their gear to work with Asterisk.

 In a restaurant environment, I'd be looking for some sort of faceplate
 overlay to make cleanup easier.


Okay great, Polycom it is then.

The 501, does it have a good speakerphone?
Something loud and clear, no distortion?
Something that would be good with quite a bit of background noise.
Do I need to lift the handset to hear the speakerphone better?  From 
pictures online it looks like the speaker is under the handset which 
doesn't look very intuitive.

I'm hoping the cooks only have to hit one button and be able to reach 
the other kitchen, no fumbling with the handset.  I guess I can remove 
the handset.

Faceplates.. interesting, a quick search on 'polycom 501 faceplate' or 
'polycom 501 stainless steel faceplate' in google doesn't come back very 
enthusiastic.  Is there such a thing?

Thank you so far for the feedback.  It's made me feel more confident and 
excited.

-Andy

In my experience, the 501 has very good speakerphone quality.  It has 4
programmable buttons so the cooks can hit one button and connect.  We have
one mounted on the wall in our computer room.  Yes, the speaker is under the
handset, but you could take the handset off and tape down the switch if
needed.  You could also cover the phone in glad wrap (except the speaker
of course).


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[asterisk-users] ADA: DOA?

2010-10-06 Thread Ken D'Ambrosio
Hey, all.  While ADA can still be downloaded, that's about all that I see.
 No development, no recent mention, and -- perhaps worst of all -- it
appears not to work properly under 64-bit systems.  So, assuming Digium's
abandoned it, are there any suggestions of alternatives?  Right now, I'm
replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly
fat client they have; if there's something slender -- roughly in the same
line as ADA -- I'd be very interested, even if it's not free.

Thanks,

-Ken


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[asterisk-users] SPA-2102 sending local IP instead of WAN IP in SIP packets

2010-10-06 Thread bruce bruce
Hi Guys,

This is such an annoying issue whenever it comes up. The sender and receive
always receive the source public IP no matter what in the IP packets but
then SIP packets go out with something like 192.168.0.20.

In this instance, a Bell Canada DSL modem is installed and I saw the
SPA-2102 register properly but only to fail later on due to sending it's LAN
IP to the Asterisk server.

So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put
the device on DMZ in the Bell Canada DSL modem and still the same issue.

I am wondering when would manufacturers finally fully comply the SIP RFC?!

I don't have the luxury to put SIP proxy, do a VPN, install a server or
anything on client site.

Diagram:

Asterisk Server = Internet = Bell Canada Modem = SPA2102

Please send me your suggestions on how to fix this if you have this type of
experience with SPA-2102

Thanks for the input,
Bruce
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Re: [asterisk-users] 2 way intercom recommendationforrestaurantkitchens

2010-10-06 Thread Andy Graybeal

 In my experience, the 501 has very good speakerphone quality.  It has 4
 programmable buttons so the cooks can hit one button and connect.  We have
 one mounted on the wall in our computer room.  Yes, the speaker is under the
 handset, but you could take the handset off and tape down the switch if
 needed.  You could also cover the phone in glad wrap (except the speaker
 of course).


Is there a Polycom 501 that is POE and one that isn't?  Or all they all POE?

-Andy


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Re: [asterisk-users] 2 wayintercom recommendationforrestaurantkitchens

2010-10-06 Thread Danny Nicholas


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 wayintercom
recommendationforrestaurantkitchens


 In my experience, the 501 has very good speakerphone quality.  It has 4
 programmable buttons so the cooks can hit one button and connect.  We have
 one mounted on the wall in our computer room.  Yes, the speaker is under
the
 handset, but you could take the handset off and tape down the switch if
 needed.  You could also cover the phone in glad wrap (except the speaker
 of course).


Is there a Polycom 501 that is POE and one that isn't?  Or all they all POE?

-Andy

The one's we use aren't.  That may be an option.


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Re: [asterisk-users] 2 way intercom recommendationforrestaurantkitchens

2010-10-06 Thread Mike
Polycom 501s were designed before the PoE standards were set in stone.  So
the PoE is actually not part of the phone, but part of the special PoE cable
that is optional. So you absolutely need that special RJ45-like cable.

Mike

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 4:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 way intercom
recommendationforrestaurantkitchens


 In my experience, the 501 has very good speakerphone quality.  It has 
 4 programmable buttons so the cooks can hit one button and connect.  
 We have one mounted on the wall in our computer room.  Yes, the 
 speaker is under the handset, but you could take the handset off and 
 tape down the switch if needed.  You could also cover the phone in 
 glad wrap (except the speaker of course).


Is there a Polycom 501 that is POE and one that isn't?  Or all they all POE?

-Andy


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Re: [asterisk-users] AMI connection limit

2010-10-06 Thread Danny Nicholas
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Wednesday, October 06, 2010 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AMI connection limit

Is there a limit to the AMI connections?
I have upto 3 process that might connect and originate calls,
2 of those processes are very infrequent - however the main one
can issue a number of calls one right after the other.

I am seeing a message when I am logging failed manager connection 
sometimes.

Are there any limits I could be running into? It does start responding 
again with no changes.
I am running 1.4.X

Thanks,

Jerry

From what I read, AMI is somehow affected by call-limit.


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[asterisk-users] CALLERPRES() with Queue

2010-10-06 Thread Rodrigo Lang
Good afternoon list,

I'm having a problem using the function CALLERPRES() when connection to a
Queue().

When I call an extension, before the Dial (), I select the function
CALLERPRES () as unavailable to link the extension comes as anonymous. But
if I call a queue before the Queue (), I select the function CALLERPRES() as
unavailable, but the identification appears normal.

Is it a problem or configuration? Someone can have for that?


Regards,

-- 
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http://rodrigorecipes.blogspot.com/http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html
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[asterisk-users] How to learn encrypted VoIP development for embedded systems

2010-10-06 Thread Zeeshan Zakaria
Hi list,

A few times I have been asked if I could do encrypted VoIP development, for
embedded systems, and in C++. And my answer has been in negative.

Now I am thinking I should start learning how to do it, but I have no clue
where to start from. I have been developing in Java for some time now, but
haven't touched C++ in years. I haven't programmed for embedded systems.
Even if I knew C++ well enough, I have no idea how to program my own
protocols and then also come up with some encryption methods for them.

I'll appreciate if those of you who have experience in this field could
guide me to any references, links, books, or other learning sources.

Sincerely,

-- 
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[asterisk-users] integrate Intertel Axxess with Asterisk

2010-10-06 Thread marvin horst
Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone
system via a SIP trunk using the IPRC card?

-- 
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Re: [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

2010-10-06 Thread Dennison Williams
On 09/22/2010 08:36 AM, Carlos Chavez wrote:
 Do you have a localnet statement in your sip.conf?  That and using
 nat=no will make sure Asterisk does not replace the IP address in the
 Invite.
   

I just wanted to give a +1 for this response.  I am using openvpn to
connect road warriors and remote offices to a central asterisk server. 
When setting up the configuration for the road warriors I created a new
subnet for them, but forgot to include their subnet as a localnet
directive in sip.conf.  The result was that sip clients on the road
warrior network would be able to register, but then when initiating a
sip call the 200 response (to the INVITE from the client) from the
asterisk server would include a contact address for some external ip
that I did not recognize.  This hint here allowed me to fix this bug,
now calls from the road warrior subnet are coming in fine.  Thanks!

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Re: [asterisk-users] SPA-2102 sending local IP instead of WAN IP in SIP packets

2010-10-06 Thread Kyle Kienapfel
On Wed, Oct 6, 2010 at 12:50 PM, bruce bruce bruceb...@gmail.com wrote:

 Hi Guys,

 This is such an annoying issue whenever it comes up. The sender and receive
 always receive the source public IP no matter what in the IP packets but
 then SIP packets go out with something like 192.168.0.20.

 In this instance, a Bell Canada DSL modem is installed and I saw the
 SPA-2102 register properly but only to fail later on due to sending it's LAN
 IP to the Asterisk server.

 So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put
 the device on DMZ in the Bell Canada DSL modem and still the same issue.

 I am wondering when would manufacturers finally fully comply the SIP RFC?!

 I don't have the luxury to put SIP proxy, do a VPN, install a server or
 anything on client site.

 Diagram:

 Asterisk Server = Internet = Bell Canada Modem = SPA2102

 Please send me your suggestions on how to fix this if you have this type of
 experience with SPA-2102

 Thanks for the input,
 Bruce


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Are you using stun?
http://en.wikipedia.org/wiki/Session_Traversal_Utilities_for_NAT
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Re: [asterisk-users] How to test BRI lines energy saving mode ?

2010-10-06 Thread Lyle Giese
Olivier wrote:
 Hello,

 If my understanding is correct, these days it seems that many ISDN BRI
 lines are configured in energy saving mode in which signalling
 D-channel is dropped until a new call comes in.

 Is it possible to replicate this behaviour with Asterisk (when
 Asterisk is in NT mode and is seen as a public ISDN by another PBX,
 for instance) ?
 If not, would you it would be a useful addition to Asterisk ?

 Regards


Energy saving???  I don't think so. 

If the D channel is down, how would I make an outgoing phone call? 
Something in this mode or your explanation just does not sound right...

Lyle Giese
LCR Computer Services, Inc.


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Re: [asterisk-users] DISA does not accept pause from cellphones when upgrading from 1.4 to 1.6

2010-10-06 Thread Matt Riddell
On 5/10/10 8:17 AM, Alejandro Recarey wrote:
 I just upgraded my asterisk box from 1.4 + Zaptel to 1.6 + DAHDI and
 services I was using perfectly before are suddenly broken.

 I have a DISA access configured, and my companies employees use if to
 dial into the companies extension from their cell phones.

 For example they would dial DISA-ACCESS-NUMBER(pause)EXTENSION.
 Has anybody else experienced this problem? Any tip would be welcome.

Nope, but maybe you should try a double pause?  Also, maybe try enabling 
DTMF logging in /etc/asterisk/logger.conf and doing a logger reload in 
the console?

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