[Asterisk-Users] Upgrading *

[[EMAIL PROTECTED]]

What is the recommended method for upgrading/downgrading from one 
version to another (after I've downloaded the new sources)?  Is it just:


make clean
make
make install

I've checked the wiki and I don't really see any mention of the 
preferred way to do it.  Thanks.


Peder
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[Asterisk-Users] Upgrading

[Chris Blunt]

Hi List, 

 

I was wondering what is the best way to upgrade an Asterisk
system to the latest version.

 

I know there is the patch method, but if I am jumping 3 or 4
versions is a re-install the best way?

 

Should I just make the files then manually copy them in? 
Does this avoid overwriting any modified sound files etc?  Should I delete the
current files or move / make a copy to a different location first?

 

I know this is a lot of questions but I am hoping for a best
practice idea etc…

 

Regards

 

Chris

 

--

 

Chris Blunt

Entropy IT Ltd

 






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[Asterisk-Users] Upgrading Asterisk

[Martin Roy]

I'm about to upgrade my currently running asterisk server from 1.0.3 to 
1.0.6 is there anything I should do before doing the upgrade?

I know since I'm using Fedora Core 3 that I must do for Zaptel : make 
clean then make linux26 and finally make install.

Do I have to remove the version 1.0.3 first or only doing install will 
replace all existing modules without removing my config files?

Once I have installed Zaptel 1.0.6 then do I have to do something 
special to upgrade asterisk from 1.0.3 to 1.0.6?

As I have a lot of people using the asterisk server I can't put it down 
for a long time so I want to be sure I don't make any stupid mistake 
before doing the upgrade as I don't have time to reinstall everything 
from scratch if something goes bad.

Thanks
Martin
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[asterisk-users] Upgrading sox

[René Christensen]

Hi,

I'm currently running an * version 1.2.13 and sox version 12.17.5. I want to 
upgrade sox to the newest release ( 12.18.2 ); need mp3 support.

But how do I make the upgrade.
Do I need to recompile asterisk afterwards?
If I make a " sox -h" after a reboot  I can see the new version is running 
but is that enough?


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Re: [Asterisk-Users] Upgrading

[Steven Ringwald]

Chris Blunt wrote:


Hi List,

I was wondering what is the best way to upgrade an Asterisk system to 
the latest version.


I know there is the patch method, but if I am jumping 3 or 4 versions 
is a re-install the best way?


Should I just make the files then manually copy them in? Does this 
avoid overwriting any modified sound files etc? Should I delete the 
current files or move / make a copy to a different location first?


I know this is a lot of questions but I am hoping for a best practice 
idea etc…




I believe that "make upgrade" installs just the applications, and does 
not touch config files (which are only installed with "make setup", BTW) 
and the sound files.


Hope this helps.
Steve

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[Asterisk-Users] Upgrading asterisk

[Thomas Kenyon]

Is it neccesary to upgrade Zaptel at the same time as upgrading asterisk.

For the second time now, I've had asterisk on a production machine
completely freeze (with no messages in any of the log files) and
eventually had to be kill -9'd.

The machine has a a TDM400 with 1xFXS and 3xFXO cards in it, the most
recent time this happened was the day after upgrading to Asterisk 1.2.8
(where I didn't update zaptel at the same time).

TIA for any help with this.
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[Asterisk-Users] Upgrading AAH

[Rolf Brusletto]

All - I've a new system, that since it's been in production, has seen a 
few issues, that look like they should be fixed by upgrading asterisk @ 
home to the latest version. I was curious if anybody out there can tell 
me their experiences with this, and what to expect.


Thanks,

Rolf Brusletto
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[Asterisk-Users] Upgrading Queue App

[John Congdon]

I have something weird happening and maybe I just need another pair
of eyes to see it.
I have been using the queue app from CVS 06/06/2003 for a couple of
months not without a problem.
Every time I upgrade (Which was difficult because of my own errors),
my customers get put in the queue just fine.  My Agents log in just
fine.  But once they get connected, there is no sound.
If I setup a simple extension:
exten => 515,1,Dial,Agent/200|15|t
I can dial 515 and talk to the agent just fine.
But if it connects through the queue app, I can't.
Anyone have any idea?

agents.conf

[agents]
ackcall => no
wrapuptime => 0
agent => 200,,Test

queues.conf
[reception]
context = incoming
strategy = ringall
music = default
timeout = 15
retry = 5
member => Agent/200
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[asterisk-users] Upgrading to 1.4

[lotusscript]

A good while back when installing 1.2 there were major issues with UK 
callerid.  Asterisk 1.2 didn't recognise the callerid correctly because 
of the way BT sent the information.  Sometimes before the first ring or 
just after.  After applying a third party patch we got it to work.  We 
were afraid to touch it after that  :-)  Has this problem now gone away 
with 1.4?

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[asterisk-users] Upgrading to 1.4

[lotusscript]

A good while back when installing 1.2 there were major issues with UK
callerid.  Asterisk 1.2 didn't recognise the callerid correctly because
of the way BT sent the information.  Sometimes before the first ring or
just after.  After applying a third party patch we got it to work.  We
were afraid to touch it after that  :-)  Has this problem now gone away
with 1.4?


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[asterisk-users] Upgrading to 1.4

[lotusscript]

A good while back when installing 1.2 there were major issues with UK
callerid.  Asterisk 1.2 didn't recognise the callerid correctly because
of the way BT sent the information.  Sometimes before the first ring or
just after.  After applying a third party patch we got it to work.  We
were afraid to touch it after that  :-)  Has this problem now gone away
with 1.4?



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Re: [asterisk-users] Upgrading sox

[Tzafrir Cohen]

On Tue, Nov 07, 2006 at 11:14:43AM +0100, René Christensen wrote:
> Hi,
> 
> I'm currently running an * version 1.2.13 and sox version 12.17.5. I want 
> to upgrade sox to the newest release ( 12.18.2 ); need mp3 support.
> But how do I make the upgrade.
> Do I need to recompile asterisk afterwards?

No. Asterisk is not linked with sox.

> If I make a " sox -h" after a reboot  I can see the new version is running 
> but is that enough?

A reboot should not be needed either.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [Asterisk-Users] Upgrading asterisk

[Doug Lytle]

Thomas Kenyon wrote:

Is it neccesary to upgrade Zaptel at the same time as upgrading asterisk.

  


I do as a matter or course.  Libpri, Zaptel, Asterisk, Asterisk-addons 
and Sounds.


Doug

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Re: [Asterisk-Users] Upgrading asterisk

[Thomas Kenyon]

Doug Lytle wrote:
> Thomas Kenyon wrote:
>> Is it neccesary to upgrade Zaptel at the same time as upgrading
>> asterisk.
>>
>>   
>
> I do as a matter or course.  Libpri, Zaptel, Asterisk, Asterisk-addons
> and Sounds.
>
> Doug
>
The problem with zaptel is that even if you can unload the modules and
reload them again, it still involves some downtime.

Will look at doing it over the weekend (unless I get another crash.)
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>

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[Asterisk-Users] Upgrading to 1.2.5?

[Martin Joseph]

Probably just me being dumb,  but I am trying to update my asterisk to 
the latest (1.2.5)


When I go to my /usr/src/asterisk  I type:

make upgrade
make install

This seems to be doing it's thing, but when I type show version from 
the console (after a restart) I still get:


Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on 
a Power Macintosh running Darwin on 2006-03-04 20:48:08 UTC


This seems like the same version number I had before also the copyright 
only goes through 2005?


Thanks for your help.

Marty

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Re: [Asterisk-Users] Upgrading AAH

[john]

With pen in hand, Rolf Brusletto succussfully stormed bulwarks which
others armed with sword and excommunication have been repulsed, and said
...
> All - I've a new system, that since it's been in production, has seen a
> few issues, that look like they should be fixed by upgrading asterisk @
> home to the latest version. I was curious if anybody out there can tell
> me their experiences with this, and what to expect.
>

Rolf,

I upgraded from 2.2 to 2.4 with only minor issues aferwards.

Back up your /etc/asterisk directory before you do anything, of course,
then untar asteriskathome.tar.gz distro in /var/aah_load directory and run
the install.sh script.

If you use sugar crm then back this up too because it overwrites
everything. You will also have to reset all the passwords as these are set
back to the standard initial passwords that [EMAIL PROTECTED] sets up.

After you're done with the upgrade, just diff all the etc/asteisk conf
files and also check, through the amp portal, all your settings. A few
parts on mine disappeared, but I had saved all the configs so I think my
total time to rebuild and reset everything was under an hour.

Of course, I forgot about sugar, which I use, but not enough to have
remembered to back it up, but that's another issue.

Hope this helps answers your question.

Regards,

John C.


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Re: [Asterisk-Users] Upgrading asterisk

[Thomas Kenyon]

Thomas Kenyon wrote:
> Doug Lytle wrote:
>   
>> Thomas Kenyon wrote:
>> 
>>> Is it neccesary to upgrade Zaptel at the same time as upgrading
>>> asterisk.
>>>
>>>   
>>>   
>> I do as a matter or course.  Libpri, Zaptel, Asterisk, Asterisk-addons
>> and Sounds.
>>
>> Doug
>>
>> 
> The problem with zaptel is that even if you can unload the modules and
> reload them again, it still involves some downtime.
>
> Will look at doing it over the weekend (unless I get another crash.)
>   
With new zaptel, this is still happening, still can't find anything in
the log files that is relevant, it just seems to be business as usual
then freeze.

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[Asterisk-Users] Upgrading the 7960 Image

[Shawn Bowen]

I am in the process of upgrading a Cisco 7960 to the P0S3-06-3-
00.bin image. It's hanging when it reaches the status message
"upgrading". I have two other phones that successfully upgraded.
When I researched on the internet it appears that there is a
problem with upgrading directly from the image that is currently
on this particular phone, PC030300, and that I should go to
P0S30203.bin first.
So far, so good. But, I don't have PC030300 and I'm finding it
impossible to get the $8 support contract since I'm in the US. I
tried google'ing on it with no luck. Is there someone out there
who has this file that would be willing to email it to me?
Any help will be very appreciated.
Best,
Shawn
[EMAIL PROTECTED]
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Re: [asterisk-users] Upgrading to 1.4

[Rob Hillis]

As is just about always the case, posting twice to the list within three 
hours is not only unlikely to get a faster response, I would in fact 
imagine it would /reduce/ your chances of getting a response at all.


lotusscript wrote:

A good while back when installing 1.2 there were major issues with UK
callerid.  Asterisk 1.2 didn't recognise the callerid correctly because
of the way BT sent the information.  Sometimes before the first ring or
just after.  After applying a third party patch we got it to work.  We
were afraid to touch it after that  :-)  Has this problem now gone away
with 1.4?
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Re: [asterisk-users] Upgrading to 1.4

[Gordon Henderson]

On Sat, 26 Apr 2008, Rob Hillis wrote:

> As is just about always the case, posting twice to the list within three 
> hours is not only unlikely to get a faster response, I would in fact imagine 
> it would /reduce/ your chances of getting a response at all.

I suspect he didn't. I've seen many instances here where posts appear 
twice (or more). In my own experience of running mailling lists over the 
years, I've found this is often caused by some broken Exchnge server 
somewhere

> lotusscript wrote:
>> A good while back when installing 1.2 there were major issues with UK
>> callerid.  Asterisk 1.2 didn't recognise the callerid correctly because
>> of the way BT sent the information.  Sometimes before the first ring or
>> just after.  After applying a third party patch we got it to work.  We
>> were afraid to touch it after that  :-)  Has this problem now gone away
>> with 1.4?

No idea about this though - I had the same issues, but have stuck to 1.2 
for the time being.

(And from what I recall, it was nothing to do with the way BT sent the 
information, but the wctdm driver simply being "broke" until the patch 
fixed it)

Gordon


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Re: [asterisk-users] Upgrading to 1.4

[Matt Brown]

On 25 Apr 2008, at 15:58, lotusscript wrote:

> A good while back when installing 1.2 there were major issues with UK
> callerid.  Asterisk 1.2 didn't recognise the callerid correctly  
> because
> of the way BT sent the information.  Sometimes before the first ring  
> or
> just after.  After applying a third party patch we got it to work.  We
> were afraid to touch it after that  :-)  Has this problem now gone  
> away
> with 1.4?

I run Asterisk 1.4.19 with Zaptel 1.4.10 on Ubuntu with a TDM400P -  
and still having issues with Callerid and Distinctive Ring.

The only way I have managed to get callerid to work with success is to  
patch the Zaptel source in 1.4.5.1 - however distinctive ring is  
broken - but this appears to be a Chan_Zap issue rather than Zaptel.

Regards

Matt

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Re: [asterisk-users] Upgrading to 1.4

[Phil Reynolds]

On Mon, Apr 28, 2008 at 01:17:33PM +0100, lotusscript wrote:
> 
> A good while back when installing 1.2 there were major issues with UK
> callerid.  Asterisk 1.2 didn't recognise the callerid correctly because
> of the way BT sent the information.  Sometimes before the first ring or
> just after.  After applying a third party patch we got it to work.  We
> were afraid to touch it after that  :-)  Has this problem now gone away
> with 1.4?

I am on an unpatched 1.2 and BT caller ID works fine for me. However,
that is not to say there may be peculiar practice elsewhere.

What won't work is distinctive ring detection and caller ID together.

-- 
Phil Reynolds
 o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95

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Re: [Asterisk-Users] Upgrading to 1.2.5?

[Kristian Kielhofner]

Martin Joseph wrote:


Probably just me being dumb,  but I am trying to update my asterisk to 
the latest (1.2.5)


When I go to my /usr/src/asterisk  I type:

make upgrade
make install

This seems to be doing it's thing, but when I type show version from the 
console (after a restart) I still get:


Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on a 
Power Macintosh running Darwin on 2006-03-04 20:48:08 UTC


This seems like the same version number I had before also the copyright 
only goes through 2005?


Thanks for your help.

Marty


Marty,

Try "make update" and "make upgrade".

--
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Re: [Asterisk-Users] Upgrading to 1.2.5?

[John Jensen]

I think you need to:
- pull the 1.2.5 tar.gz file from ftp/http
- extract it into a dir (tar xvfz filename)
- cd into it
- excecute make upgrade and make install

Cheers,

John

>>> [EMAIL PROTECTED] 04-03-06 21:12 >>>
Probably just me being dumb,  but I am trying to update my asterisk to 
the latest (1.2.5)

When I go to my /usr/src/asterisk  I type:

make upgrade
make install

This seems to be doing it's thing, but when I type show version from 
the console (after a restart) I still get:

Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on 
a Power Macintosh running Darwin on 2006-03-04 20:48:08 UTC

This seems like the same version number I had before also the copyright 
only goes through 2005?

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[asterisk-users] Upgrading DAHDI and Asterisk

[Alex Saavedra]

Hello,

I just upgraded DAHDI from 2.3.0 to 2.4.0, both dahdi-linux and dahdi-tools
(Ubuntu 10.04 64 bits). It took a few minutes, and it was straightforward.
Everything is working.

Now I'm preparing to upgrade Asterisk from 1.6.2.7 to 1.6.2.15. I made a
backup of configuration files, codec licenses and CDR. Is there something
else I should be aware of before upgrading?

Thank you,

Alex Saavedra
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[Asterisk-Users] Upgrading asterisk yields broken pipe

[jimmy . gatt]

Hello list,

I am attempting to upgrade asterisk on a production box.  I have opted to set 
INSTALL_PREFIX to /usr/local/asterisk-0.7.2 which is ugly (since it 
makes /usr, /var, /etc directories in there), but I didn't want the new install 
to overwrite my existing installation.

The new asterisk runs, but when a call tries to go through, I get six 
of "Ouch ... error while writing audio data: : Broken pipe" errors, and then a 
segfault.  Reading elsewhere, I have discovered that the error is coming from 
mpg123.  The process table yields 6 mpg123 processes which appear to be playing 
the on hold music ("onhold_low.mp3", "onhold_high.mp3", etc.)  Presently, I 
don't know where to go.

1. I am trying to test a new compile of asterisk while retaining the ability to 
revert to the old compilation.  Am I going about it the right way (by setting 
the INSTALL_PREFIX to a different dir)?

2. If the answer to #1 is "yes", then what might be the problem with mpg123?

-
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[Asterisk-Users] Upgrading my HOP-1002 software

[Mohamed M Moustafa]

Hi,

I need to upgrade my HOP-1002 ip phones (i am currently running
ver.1.35), i need the source of the latest software and the
steps to do it.

Thanks in advance.

I am also configuring Asterisk as my SIP server, any
recommendations ?

Regards,
Mohammed Mahmoud
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[asterisk-users] upgrading from A101 to....A102

[Bill Gibbs]

Any benefit on getting the PCI Express version?

 

Bill

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[asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2

[John A. Sullivan III]

Hello, all.  After reading the README, UPGRADE.txt, and a quick tour
through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
one simply compiles and installs over the old installation being careful
to NOT install the sample files? Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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[asterisk-users] Upgrading Asterisk in Trixbox installation

[ilker Aktuna]

Hi,

My Trixbox 2.8.0.1 installation includes the following Asterik version:
1.6.0.9-samy-r27

I am having some problems with it and I think they might be solved if I use the 
latest Asterisk version.
Is it a good idea to update Asterisk in Trixbox externally ?
Is it safe ?

If so, which version should I prefer ?
1.6.1.5 or 1.6.0.14 ?

Thanks,
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[Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

[Chris Bagnall]

Hello all,

I'm contemplating upgrading a client's asterisk system from 1.0.9 to 1.2
beta to take advantage of some of the new echo cancellers in the later
zaptel packages. Problem is, I'll be doing it without physical access to the
box and without being able to personally test the new echo cancellation for
them, so I'll be relying on information they provide me with.

Their setup involves a Rev. I TDM400 card with 3 FXO modules connected to
standard BT analogue lines. They've been complaining about echo for some
time, despite the multitude of options I've tried in zapata.conf to limit
the echo problem.

Here are the current zapata.conf settings:
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=12.0
txgain=8.0

(rxgain and txgain calculated by running ztmonitor on a number of different
calls over a period of a few days, aiming to keep the levels in the middle)

1) Is an upgrade to 1.2 likely to help at all?

2) If yes, which echo canceller is most likely to yield favourable results,
and are there any changes I should make in the conf file?

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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[asterisk-users] Upgrading from 1.0.9 to 1.2.6

[Matt]

Hi,
I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk
version.   What do I need to be aware of?  I AM aware 1.2.6 is not the
newest version, but anything above .6, at this time, seems to have
stability issues (I've tried them on multiple machines)
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[asterisk-users] Upgrading from 1.2.12.1 to 1.2.13

[Henrik Woffinden]

Hi,

After upgrading from:
Zaptel 1.2.9.1
Asterisk 1.2.12.1 with bristuff-0.3.0-PRE-1s
to
Zaptel 1.2.10
Asterisk 1.2.13 with brustuff-0.3.0-PRE-1v

I get the following error when connecting my Xlite Softphone:
--- cut ---
Nov  4 17:33:45 WARNING[4430]: chan_sip.c:1090 __sip_xmit: sip_xmit of
0x886df58 (len 486) to 192.168.9.9:31308 returned -1: Operation not
permitted
--- cut ---

It seems to be Xlite wanting to see who of my contacts is on-line.
There's no problem phoning, but all my contacts are "offline" according
to Xlite.
"sip show peers" on the CLI tells me different. There are hint lines for
everybody. And it worked perfectly in 1.2.12.1

Does anyone know what the error could be?

-- 
Med venlig hilsen / Best regards,

Henrik Woffinden


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[asterisk-users] Upgrading my office - Need help

[Gary Guthary]

Hi -

If I'm posting this in the wrong place, pease forgive me

Folks, I need help...

One company I consult for is upgrading their office and will need a PBX
replacement in the next two months.

I'm seriously thinking of offering them an 'Asterisk' solution versus them
getting locked in with some PBX vendor.

This means I've got to come up with some sort of demo system to show them.

I've got the hardware. - Dedicated Linux box I can re-config and/or re-load
as needed. - Have one FXS/FXO card to demo intfc to telco. - And a bunch of
Cisco 79xx phones I can use for demos.

If I can get this rolling, I'll be more than happy to pay ***MONEY*** to
anybody who can help. - Also, I'm not afraid to pay for "already-developed"
admin software I can use to manage my system. - I just don't know 'which is
which'.

To prevent cluttering up this board, please send all responses to:
[EMAIL PROTECTED]

Thanks in advance & again apologize if this is not the right place to post
this.

Gary Guthary


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[asterisk-users] upgrading from 1.4.39 to 1.8.5

[Joseph]

What sort of things should I watch out for when upgrading from 1.4.39 to 1.8.5

Thanks,
--
Joseph

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[asterisk-users] Upgrading from 1.4 to 11.4.0

[Eric Smith]

Hi

After upgrading from 1.4 to 11.4.0, I am able to receive calls
and direct them to extensions via defined trunks.

However, when making outgoing calls I receive the following
error:

-- Executing [00044111@default:4] Dial("SIP/fixedline-0004", 
"SIP/mydevice/0044111,60,w") in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/mydevice/0044111
 WARNING[13053][C-0002]: channel.c:6164 ast_channel_make_compatible_helper: 
No path to translate from SIP/mydevice-0005 to SIP/hardphone-0004

And when I try to try to initiate a call with a manager script, I receive an 
authentication error from the script.

How might I find more info to help diagnose either or both of these issues?

-- 
Eric Smith

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Re: [Asterisk-Users] Upgrading my HOP-1002 software

[Gary]

On Sun, 29 May 2005 14:51:08 GMT, Mohamed M Moustafa wrote:

>
>Hi,
>
>I need to upgrade my HOP-1002 ip phones (i am currently running
>ver.1.35), i need the source of the latest software and the
>steps to do it.

www.aredfox.com

>Thanks in advance.
>
>I am also configuring Asterisk as my SIP server, any
>recommendations ?
>
>Regards,
>Mohammed Mahmoud
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Re: [asterisk-users] upgrading from A101 to....A102

[Michiel van Baak]

On 16:21, Thu 22 Feb 07, Bill Gibbs wrote:
> Any benefit on getting the PCI Express version?

If you have a PCI Express slot, go for it.
The hardware EC is more important to me, but the machines I
use them in have normal PCI slots so that's why I dont use
the PCI Express version yet. Prolly in my next machine...

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [asterisk-users] upgrading from A101 to....A102

[Tom]

At 03:21 PM 2/22/2007, you wrote:

Content-Type: multipart/alternative;
boundary="_=_NextPart_001_01C756C7.503C75EE"
Content-class: urn:content-classes:message

Any benefit on getting the PCI Express version?


Yes but only if you are short on PCI slots and have a spare PCIe port 
that you want to donate.


Otherwise no functional benefits.  Only physical.

Tom



Bill
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RE: [asterisk-users] upgrading from A101 to....A102

[Porier, Jeremy M.]

We're having a lot of D channel problems with the pci-e on HP servers.
Going to PCI fixed the problem.  Sangoma is aware of the problem and is
using one of our servers to work toward a solution.
 
-Jeremy



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: Thursday, February 22, 2007 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] upgrading from A101 toA102



Any benefit on getting the PCI Express version?

 

Bill

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Re: [asterisk-users] upgrading from A101 to....A102

[Jorge de Diego]

Hi Jeremy,

We had D channels problems with A102De (A102 with HWEC and PCI-Express
version), and it was solved from Sangoma changing one parameter in
wanpipe.conf.

We have HP server too in this installation.

Our problem with D-channel was when wanted use only half-E1 channels (really
we continue having 15 channels up from telco), and we wanted limit them in
wanpipe config.

Here show you our wanpipe.conf:

[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE   = AFT
S514CPU = A
CommPort= PRI
AUTO_PCISLOT= NO
PCISLOT = 4
PCIBUS  = 14
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= NCRC4
FE_LINE = 1
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
TE_HIGHIMPEDANCE= NO
LBO = 120OH
TE_SIG_MODE = CCS
FE_TXTRISTATE   = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN   = 1
TDMV_DCHAN  = NO

[w1g1]
ACTIVE_CH   = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC   = YES

Our change was set ACTIVE_CH = ALL and every sync problems with telco about
D-channels was solved.

Hope this helps you.

Regards



On 23/2/07 17:16, "Porier, Jeremy M." <[EMAIL PROTECTED]> wrote:

> We're having a lot of D channel problems with the pci-e on HP servers.  Going
> to PCI fixed the problem.  Sangoma is aware of the problem and is using one of
> our servers to work toward a solution.
>  
> -Jeremy
> 
> 
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
> Sent: Thursday, February 22, 2007 2:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] upgrading from A101 toA102
> 
> Any benefit on getting the PCI Express version?
>  
> Bill
> 
> 
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Re: [asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2

[John A. Sullivan III]

On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> Hello, all.  After reading the README, UPGRADE.txt, and a quick tour
> through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
> one simply compiles and installs over the old installation being careful
> to NOT install the sample files? Thanks - John

We've hit a problem even before installing.  We're using Zimbra as IMAP
storage for our voicemails.  When we run make menuselect in 1.6.1.2, the
IMAP storage option is disabled (XXX).  When we run menuselect in
1.6.1.1 on the same system, it is available and enabled.  Did we miss
something or is this a bug? Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2

[Tilghman Lesher]

On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> > Hello, all.  After reading the README, UPGRADE.txt, and a quick tour
> > through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
> > one simply compiles and installs over the old installation being careful
> > to NOT install the sample files? Thanks - John
>
> We've hit a problem even before installing.  We're using Zimbra as IMAP
> storage for our voicemails.  When we run make menuselect in 1.6.1.2, the
> IMAP storage option is disabled (XXX).  When we run menuselect in
> 1.6.1.1 on the same system, it is available and enabled.  Did we miss
> something or is this a bug? Thanks - John

Re-run configure on 1.6.1.1.  It's likely that the option will go away, as the
dependency is no longer met.

-- 
Tilghman

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Re: [asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2

[John A. Sullivan III]

On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote:
> On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> > On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> > > Hello, all.  After reading the README, UPGRADE.txt, and a quick tour
> > > through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
> > > one simply compiles and installs over the old installation being careful
> > > to NOT install the sample files? Thanks - John
> >
> > We've hit a problem even before installing.  We're using Zimbra as IMAP
> > storage for our voicemails.  When we run make menuselect in 1.6.1.2, the
> > IMAP storage option is disabled (XXX).  When we run menuselect in
> > 1.6.1.1 on the same system, it is available and enabled.  Did we miss
> > something or is this a bug? Thanks - John
> 
> Re-run configure on 1.6.1.1.  It's likely that the option will go away, as the
> dependency is no longer met.
> 
I'm not sure I understand.  Nothing has changed to make the dependency
fail.  This is the same device where we are quite successfully running
voicemail in IMAP - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2

[John A. Sullivan III]

On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote:
> On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote:
> > On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> > > On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> > > > Hello, all.  After reading the README, UPGRADE.txt, and a quick tour
> > > > through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
> > > > one simply compiles and installs over the old installation being careful
> > > > to NOT install the sample files? Thanks - John
> > >
> > > We've hit a problem even before installing.  We're using Zimbra as IMAP
> > > storage for our voicemails.  When we run make menuselect in 1.6.1.2, the
> > > IMAP storage option is disabled (XXX).  When we run menuselect in
> > > 1.6.1.1 on the same system, it is available and enabled.  Did we miss
> > > something or is this a bug? Thanks - John
> > 
> > Re-run configure on 1.6.1.1.  It's likely that the option will go away, as 
> > the
> > dependency is no longer met.
> > 
> I'm not sure I understand.  Nothing has changed to make the dependency
> fail.  This is the same device where we are quite successfully running
> voicemail in IMAP - John

Very strange, I did as you suggested and sure enough, IMAP is disabled
as an option in 1.6.1.1.  I'll have to dig a little deeper as I believe
the most we may have done was a yum update! Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2

[John A. Sullivan III]

On Mon, 2009-08-03 at 14:27 -0400, John A. Sullivan III wrote:
> On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote:
> > On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote:
> > > On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> > > > On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> > > > > Hello, all.  After reading the README, UPGRADE.txt, and a quick tour
> > > > > through google, is it safe to assume to upgrade from 1.6.1.1 to 
> > > > > 1.6.1.2,
> > > > > one simply compiles and installs over the old installation being 
> > > > > careful
> > > > > to NOT install the sample files? Thanks - John
> > > >
> > > > We've hit a problem even before installing.  We're using Zimbra as IMAP
> > > > storage for our voicemails.  When we run make menuselect in 1.6.1.2, the
> > > > IMAP storage option is disabled (XXX).  When we run menuselect in
> > > > 1.6.1.1 on the same system, it is available and enabled.  Did we miss
> > > > something or is this a bug? Thanks - John
> > > 
> > > Re-run configure on 1.6.1.1.  It's likely that the option will go away, 
> > > as the
> > > dependency is no longer met.
> > > 
> > I'm not sure I understand.  Nothing has changed to make the dependency
> > fail.  This is the same device where we are quite successfully running
> > voicemail in IMAP - John
> 
> Very strange, I did as you suggested and sure enough, IMAP is disabled
> as an option in 1.6.1.1.  I'll have to dig a little deeper as I believe
> the most we may have done was a yum update! Thanks - John
Ah, I remember now and shame on us for not documenting it.  I'll record
it here in case someone else hits the same thing.

The version of libc-client that ships with CentOS 5.3 is too old to work
with Zimbra. We thus needed to use the later imap-2007e version.
configure needs to point to it, in our case:
./configure  --with-imap=/home/compuser/Asterisk/imap-2007e

I'll sheepishly add that to our internal documentation now :-(
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2

[John A. Sullivan III]

On Mon, 2009-08-03 at 14:52 -0400, John A. Sullivan III wrote:
> On Mon, 2009-08-03 at 14:27 -0400, John A. Sullivan III wrote:
> > On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote:
> > > On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote:
> > > > On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote:
> > > > > On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> > > > > > Hello, all.  After reading the README, UPGRADE.txt, and a quick tour
> > > > > > through google, is it safe to assume to upgrade from 1.6.1.1 to 
> > > > > > 1.6.1.2,
> > > > > > one simply compiles and installs over the old installation being 
> > > > > > careful
> > > > > > to NOT install the sample files? Thanks - John
> > > > >
> > > > > We've hit a problem even before installing.  We're using Zimbra as 
> > > > > IMAP
> > > > > storage for our voicemails.  When we run make menuselect in 1.6.1.2, 
> > > > > the
> > > > > IMAP storage option is disabled (XXX).  When we run menuselect in
> > > > > 1.6.1.1 on the same system, it is available and enabled.  Did we miss
> > > > > something or is this a bug? Thanks - John
> > > > 
> > > > Re-run configure on 1.6.1.1.  It's likely that the option will go away, 
> > > > as the
> > > > dependency is no longer met.
> > > > 
> > > I'm not sure I understand.  Nothing has changed to make the dependency
> > > fail.  This is the same device where we are quite successfully running
> > > voicemail in IMAP - John
> > 
> > Very strange, I did as you suggested and sure enough, IMAP is disabled
> > as an option in 1.6.1.1.  I'll have to dig a little deeper as I believe
> > the most we may have done was a yum update! Thanks - John
> Ah, I remember now and shame on us for not documenting it.  I'll record
> it here in case someone else hits the same thing.
> 
> The version of libc-client that ships with CentOS 5.3 is too old to work
> with Zimbra. We thus needed to use the later imap-2007e version.
> configure needs to point to it, in our case:
> ./configure  --with-imap=/home/compuser/Asterisk/imap-2007e
> 
> I'll sheepishly add that to our internal documentation now :-(
Would anyone mind answering the original question, though.  Is it
correct to simply compile and install over 1.6.1.1 to upgrade to
1.6.1.2? Thanks - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2

[Jared Smith]

On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote:
> Hello, all.  After reading the README, UPGRADE.txt, and a quick tour
> through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2,
> one simply compiles and installs over the old installation being careful
> to NOT install the sample files?

Yes, that's a safe assumption to make, given the fact that you're just
bumping minor releases on the same development branch.

If you were moving from the 1.6.1 branch to the 1.6.2 branch, for
example, you'd definitely want to check UPGRADE.txt for more details of
configuration options that might have changed, etc.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Upgrading Asterisk in Trixbox installation

[Jeff LaCoursiere]

On Mon, 31 Aug 2009, ilker Aktuna wrote:

> Hi,
>
> My Trixbox 2.8.0.1 installation includes the following Asterik version:
> 1.6.0.9-samy-r27
>
> I am having some problems with it and I think they might be solved if I use 
> the latest Asterisk version.
> Is it a good idea to update Asterisk in Trixbox externally ?

I've done it in the 1.4 branch.

> Is it safe ?
>

Should be, as long as you stay within the same branch.  That being the 
case, I would stick with 1.6.0.14 if I were you.  Make sure you don't 
"make samples" :)

j

> If so, which version should I prefer ?
> 1.6.1.5 or 1.6.0.14 ?
>
> Thanks,
> ilker

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Re: [asterisk-users] Upgrading Asterisk in Trixbox installation

[ilker Aktuna]

Thank you.
That was quick and helpful :)

Then I'll just "make" and "make install"
What should I backup, in case of rollback requirement ?

Thanks.


- Original Message - 
From: "Jeff LaCoursiere" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Monday, August 31, 2009 11:15 PM
Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation


>
> On Mon, 31 Aug 2009, ilker Aktuna wrote:
>
>> Hi,
>>
>> My Trixbox 2.8.0.1 installation includes the following Asterik version:
>> 1.6.0.9-samy-r27
>>
>> I am having some problems with it and I think they might be solved if I 
>> use the latest Asterisk version.
>> Is it a good idea to update Asterisk in Trixbox externally ?
>
> I've done it in the 1.4 branch.
>
>> Is it safe ?
>>
>
> Should be, as long as you stay within the same branch.  That being the
> case, I would stick with 1.6.0.14 if I were you.  Make sure you don't
> "make samples" :)
>
> j
>
>> If so, which version should I prefer ?
>> 1.6.1.5 or 1.6.0.14 ?
>>
>> Thanks,
>> ilker
>
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Re: [asterisk-users] Upgrading Asterisk in Trixbox installation

[Jeff LaCoursiere]

On Mon, 31 Aug 2009, ilker Aktuna wrote:

> Thank you.
> That was quick and helpful :)
>
> Then I'll just "make" and "make install"
> What should I backup, in case of rollback requirement ?

That's a bit tougher.  At the least /usr/lib/asterisk/modules, 
/etc/asterisk, and /usr/sbin/asterisk...  someone else may need to chime 
in here...

I've always been a fan of trixbox, and I have done a lot of installations, 
but when it comes down to it all I really want it for is for a quick 
installations of asterisk and FreePBX.  I don't think I actually use any 
of the trixbox-only features.  I've also been enamored with Ubuntu of 
late, and have dumped CentOS.  YMMV, but you might consider starting over 
with a clean build of the linux of your choice, and doing asterisk + 
addons + FreePBX from source.

j

>
> Thanks.
>
>
> - Original Message -
> From: "Jeff LaCoursiere" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Monday, August 31, 2009 11:15 PM
> Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
>
>
>>
>> On Mon, 31 Aug 2009, ilker Aktuna wrote:
>>
>>> Hi,
>>>
>>> My Trixbox 2.8.0.1 installation includes the following Asterik version:
>>> 1.6.0.9-samy-r27
>>>
>>> I am having some problems with it and I think they might be solved if I
>>> use the latest Asterisk version.
>>> Is it a good idea to update Asterisk in Trixbox externally ?
>>
>> I've done it in the 1.4 branch.
>>
>>> Is it safe ?
>>>
>>
>> Should be, as long as you stay within the same branch.  That being the
>> case, I would stick with 1.6.0.14 if I were you.  Make sure you don't
>> "make samples" :)
>>
>> j
>>
>>> If so, which version should I prefer ?
>>> 1.6.1.5 or 1.6.0.14 ?
>>>
>>> Thanks,
>>> ilker
>>
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Re: [asterisk-users] Upgrading Asterisk in Trixbox installation

[Tom Moore]

One thing I kind of like that Trixbox does is their endpoint manager.
That's about the only feature I haven't been able to replace.

Tom
 



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Monday, August 31, 2009 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation


On Mon, 31 Aug 2009, ilker Aktuna wrote:

> Thank you.
> That was quick and helpful :)
>
> Then I'll just "make" and "make install"
> What should I backup, in case of rollback requirement ?

That's a bit tougher.  At the least /usr/lib/asterisk/modules, 
/etc/asterisk, and /usr/sbin/asterisk...  someone else may need to chime 
in here...

I've always been a fan of trixbox, and I have done a lot of installations, 
but when it comes down to it all I really want it for is for a quick 
installations of asterisk and FreePBX.  I don't think I actually use any 
of the trixbox-only features.  I've also been enamored with Ubuntu of 
late, and have dumped CentOS.  YMMV, but you might consider starting over 
with a clean build of the linux of your choice, and doing asterisk + 
addons + FreePBX from source.

j

>
> Thanks.
>
>
> - Original Message -
> From: "Jeff LaCoursiere" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Monday, August 31, 2009 11:15 PM
> Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
>
>
>>
>> On Mon, 31 Aug 2009, ilker Aktuna wrote:
>>
>>> Hi,
>>>
>>> My Trixbox 2.8.0.1 installation includes the following Asterik version:
>>> 1.6.0.9-samy-r27
>>>
>>> I am having some problems with it and I think they might be solved if I
>>> use the latest Asterisk version.
>>> Is it a good idea to update Asterisk in Trixbox externally ?
>>
>> I've done it in the 1.4 branch.
>>
>>> Is it safe ?
>>>
>>
>> Should be, as long as you stay within the same branch.  That being the
>> case, I would stick with 1.6.0.14 if I were you.  Make sure you don't
>> "make samples" :)
>>
>> j
>>
>>> If so, which version should I prefer ?
>>> 1.6.1.5 or 1.6.0.14 ?
>>>
>>> Thanks,
>>> ilker
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] Upgrading Asterisk in Trixbox installation

[Duncan Turnbull]

I am a big fan of ubuntu LTS and freepbx and recently I saw mention of a 
custom module to add auto configuring endpoints for linksys (but i cna't 
find it again right now)

Trixbox had too much stuff whereas the source install of just what you 
want is nice and clean

Cheers Duncan

Jeff LaCoursiere wrote:
> On Mon, 31 Aug 2009, ilker Aktuna wrote:
>
>   
>> Thank you.
>> That was quick and helpful :)
>>
>> Then I'll just "make" and "make install"
>> What should I backup, in case of rollback requirement ?
>> 
>
> That's a bit tougher.  At the least /usr/lib/asterisk/modules, 
> /etc/asterisk, and /usr/sbin/asterisk...  someone else may need to chime 
> in here...
>
> I've always been a fan of trixbox, and I have done a lot of installations, 
> but when it comes down to it all I really want it for is for a quick 
> installations of asterisk and FreePBX.  I don't think I actually use any 
> of the trixbox-only features.  I've also been enamored with Ubuntu of 
> late, and have dumped CentOS.  YMMV, but you might consider starting over 
> with a clean build of the linux of your choice, and doing asterisk + 
> addons + FreePBX from source.
>
> j
>
>   
>> Thanks.
>>
>>
>> - Original Message -
>> From: "Jeff LaCoursiere" 
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> 
>> Sent: Monday, August 31, 2009 11:15 PM
>> Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
>>
>>
>> 
>>> On Mon, 31 Aug 2009, ilker Aktuna wrote:
>>>
>>>   
>>>> Hi,
>>>>
>>>> My Trixbox 2.8.0.1 installation includes the following Asterik version:
>>>> 1.6.0.9-samy-r27
>>>>
>>>> I am having some problems with it and I think they might be solved if I
>>>> use the latest Asterisk version.
>>>> Is it a good idea to update Asterisk in Trixbox externally ?
>>>> 
>>> I've done it in the 1.4 branch.
>>>
>>>   
>>>> Is it safe ?
>>>>
>>>> 
>>> Should be, as long as you stay within the same branch.  That being the
>>> case, I would stick with 1.6.0.14 if I were you.  Make sure you don't
>>> "make samples" :)
>>>
>>> j
>>>
>>>   
>>>> If so, which version should I prefer ?
>>>> 1.6.1.5 or 1.6.0.14 ?
>>>>
>>>> Thanks,
>>>> ilker
>>>> 
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>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>   
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>> 
>
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Re: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

[Tom Rymes]

On Nov 13, 2005, at 8:36 AM, Chris Bagnall wrote:


Hello all,

I'm contemplating upgrading a client's asterisk system from 1.0.9  
to 1.2

beta to take advantage of some of the new echo cancellers in the later
zaptel packages. Problem is, I'll be doing it without physical  
access to the
box and without being able to personally test the new echo  
cancellation for

them, so I'll be relying on information they provide me with.

Their setup involves a Rev. I TDM400 card with 3 FXO modules  
connected to
standard BT analogue lines. They've been complaining about echo for  
some
time, despite the multitude of options I've tried in zapata.conf to  
limit

the echo problem.

Here are the current zapata.conf settings:
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=12.0
txgain=8.0

(rxgain and txgain calculated by running ztmonitor on a number of  
different
calls over a period of a few days, aiming to keep the levels in the  
middle)


Chris,

Before you upgrade to 1.2 and potentially break a lot of things, have  
you followed the instructions available at  to adjust the rxgain and  
txgain? I can't say without testing the machine, but those levels  
look extremely high to me , and might be exacerbating, if not causing  
the echo problem. Try to adjust the gains quantitatively using  
milliwatt lines first before you go the upgrade route.


Tom
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RE: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

[Chris Bagnall]

> Before you upgrade to 1.2 and potentially break a lot of 
> things, have you followed the instructions available at 
>  wiki/view/Asterisk+zapata+gain+adjustment> to adjust the 
> rxgain and txgain?

Don't suppose anyone knows of a 1004 Hz 0dB number I can call to test with
in the UK, do they?

(or how I might go about setting up one)

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

[Rich Adamson]

> > I'm contemplating upgrading a client's asterisk system from 1.0.9  
> > to 1.2
> > beta to take advantage of some of the new echo cancellers in the later
> > zaptel packages. Problem is, I'll be doing it without physical  
> > access to the
> > box and without being able to personally test the new echo  
> > cancellation for
> > them, so I'll be relying on information they provide me with.
> >
> > Their setup involves a Rev. I TDM400 card with 3 FXO modules  
> > connected to
> > standard BT analogue lines. They've been complaining about echo for  
> > some
> > time, despite the multitude of options I've tried in zapata.conf to  
> > limit
> > the echo problem.
> >
> > Here are the current zapata.conf settings:
> > echocancel=yes
> > echocancelwhenbridged=yes
> > echotraining=800
> > rxgain=12.0
> > txgain=8.0
> >
> > (rxgain and txgain calculated by running ztmonitor on a number of  
> > different
> > calls over a period of a few days, aiming to keep the levels in the  
> > middle)
> 
> Chris,
> 
> Before you upgrade to 1.2 and potentially break a lot of things, have  
> you followed the instructions available at  wiki/view/Asterisk+zapata+gain+adjustment> to adjust the rxgain and  
> txgain? I can't say without testing the machine, but those levels  
> look extremely high to me , and might be exacerbating, if not causing  
> the echo problem. Try to adjust the gains quantitatively using  
> milliwatt lines first before you go the upgrade route.
 
Based on my extensive professional experience and applying that
experience to the TDM card, I'd have to agree. The TDM card will
certainly generate echo with those gains.

Incrementally reduce those gains by 2db per day and listen to your
customer's feedback relative to echo. Don't bother using milliwatt
generators and ztmonitor. (Those tools are okay to find a starting
point if you have no other transmission test sets, but will not
help even one little tiny bit after that.)

You _will_ reach a point where the echo is minimal, but you will also
begin to hear complaints about 'low audio volume'. Your goal is to
find those gain values where you are trading off minimal echo with
low audio.

The TDM card does a pretty good job for those asterisk systems that are
fairly close to the central office (maybe something within about 5db
or so of pstn loss). It does a fairly poor job of balancing echo with 
proper audio levels the further one is from the CO past about 5db.

It's my opinion (and nothing more then that) that part of the problem
relates to the fairly narrow operating range of the existing echo
cancellers.  The new KB1 canceller is much better, but still no where
near commercial cancellers. About the same is true with the MG2 when
used with the TDM card. Comparing the MG2 canceller to KB1, I've seen
about another 1 db of improvement in audio levels during my early
tests of MG2. (I've stayed with the KB1 canceller for now.)

Before upgrading your customer's system to any newer version, make
sure you research "changes" to asterisk, and have a valid backout plan
in case your research misses items of importance to your customer's
operation.


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RE: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

[Chris Bagnall]

> Incrementally reduce those gains by 2db per day and listen to 
> your customer's feedback relative to echo. Don't bother using 
> milliwatt generators and ztmonitor. (Those tools are okay to 
> find a starting point if you have no other transmission test 
> sets, but will not help even one little tiny bit after that.)

Would you suggest reducing both tx and rx gain by a similar amount each day?
They've complained about them hearing echo on inbound calls, but never that
the *caller* has complained to them about echo. Does that mean txgain is OK,
or would it be worth reducing that as well?

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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RE: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

[Rich Adamson]

> > Incrementally reduce those gains by 2db per day and listen to 
> > your customer's feedback relative to echo. Don't bother using 
> > milliwatt generators and ztmonitor. (Those tools are okay to 
> > find a starting point if you have no other transmission test 
> > sets, but will not help even one little tiny bit after that.)
> 
> Would you suggest reducing both tx and rx gain by a similar amount each day?
> They've complained about them hearing echo on inbound calls, but never that
> the *caller* has complained to them about echo. Does that mean txgain is OK,
> or would it be worth reducing that as well?

My experience says the rxgain has the largest impact on echo. But, both
are important. The larger the txgain, the larger the reflected audio
tends to be, however that reflected audio is a small percentage of the
actual outbound audio, so txgain is of lessor importance.

In my case and I'm 7 db from the central office, any rxgain value above
5 will start to cause echo regardless of which canceller is used. For
my ears, that value is just a little low, but I can make up for it
by increasing the volume on the C7960 phone so its not a hugh issue at
all.

If I were in your shoes, I'd drop that rxgain from 12 down to 8 the
first day (with no other changes) and listen to your customer. Then
start the 2 db per day drop after that (both txgain and rxgain dropping
by 2 db per day).

Once your customer says things are reasonable, then you might play around
with increasing/decreasing txgain and rxgain by small amounts to see if
you can improve it any more. A lot of that either dictates that you are
on site and making test calls, or, your customer gives you honest feedback
relative to who can/cannot hear. (Your customer is not likely to give you
reasonable feedback as they won't be asking the opposite end of the call
what they hear. And, even if they did, different callers will have
different opinions based on their hearing problems, etc.)

The advantage to using a CO-based milliwatt generator (and appropriate
transmission test set) is to simply identify factually what the pstn
loss is between * and the CO. If that loss number is known, then a good
starting point for * is rxgain and txgain values about 2 db less.
(Eg, if the pstn loss is 10 db, then start with rxgain/txgain of 8.)
Many telco's will tell you exactly what the loss is to your site, so
rather then messing around with the milliwatt and transmission test
sets, just ask them. (It might take a little effort to find that person
that can tell you, but they are there and do have the numbers.)

Keep in mind that echo can be created at lots of different points, and
the above is "assuming" the echo is resulting from inappropriate gain
settings (highly likely to be the case).


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Re: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

[Rod Bacon]

I have personally done this recently, and my advice is definately DO IT.

In my situation, I noticed a marked improvement in echo and general audio 
quality.

I too had gain settings that were "out of whack" compared to what others had 
experienced, but as long as your following the documented methods (ztmonitor, 
etc.) then whatever works for you is fine.


One word of warning though, when I went to 1.2Beta2, my gain settings were all 
out of kilter again. Calls were suddenly far too quiet. I ended up setting them 
all back to 0 (you'll need to perform your tests all over again).


This may be because of the different default echo canceller in the Zaptel 
drivers? Anyway... good luck.


==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237   ICQ: 5662270
==


Chris Bagnall wrote:

Hello all,

I'm contemplating upgrading a client's asterisk system from 1.0.9 to 1.2
beta to take advantage of some of the new echo cancellers in the later
zaptel packages. Problem is, I'll be doing it without physical access to the
box and without being able to personally test the new echo cancellation for
them, so I'll be relying on information they provide me with.

Their setup involves a Rev. I TDM400 card with 3 FXO modules connected to
standard BT analogue lines. They've been complaining about echo for some
time, despite the multitude of options I've tried in zapata.conf to limit
the echo problem.

Here are the current zapata.conf settings:
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=12.0
txgain=8.0

(rxgain and txgain calculated by running ztmonitor on a number of different
calls over a period of a few days, aiming to keep the levels in the middle)

1) Is an upgrade to 1.2 likely to help at all?

2) If yes, which echo canceller is most likely to yield favourable results,
and are there any changes I should make in the conf file?

Thanks in advance.

Regards,

Chris

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Re: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

[Mojo with Horan & Company, LLC]

I'd just work on upgrading the zaptel portion, don't do asterisk until 
you're ready.  Swapping 1.0.9 for something in the 1.2.0 line went 
pretty painless for me, and asterisk couldn't care less as far as I can 
tell. I'm using 1.0.10 asterisk and 1.2.0rc2 zaptel.


Moj

Chris Bagnall wrote:

Hello all,

I'm contemplating upgrading a client's asterisk system from 1.0.9 to 1.2
beta to take advantage of some of the new echo cancellers in the later
zaptel packages. Problem is, I'll be doing it without physical access to the
box and without being able to personally test the new echo cancellation for
them, so I'll be relying on information they provide me with.

Their setup involves a Rev. I TDM400 card with 3 FXO modules connected to
standard BT analogue lines. They've been complaining about echo for some
time, despite the multitude of options I've tried in zapata.conf to limit
the echo problem.

Here are the current zapata.conf settings:
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=12.0
txgain=8.0

(rxgain and txgain calculated by running ztmonitor on a number of different
calls over a period of a few days, aiming to keep the levels in the middle)

1) Is an upgrade to 1.2 likely to help at all?

2) If yes, which echo canceller is most likely to yield favourable results,
and are there any changes I should make in the conf file?

Thanks in advance.

Regards,

Chris


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Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6

[Eric \"ManxPower\" Wieling]

Matt wrote:

Hi,
I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk
version.   What do I need to be aware of?  I AM aware 1.2.6 is not the
newest version, but anything above .6, at this time, seems to have
stability issues (I've tried them on multiple machines)


/path/to/src/asterisk/docs/UPGRADE.txt or similar file name.

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Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6

[Matt]

Thanks for the suggestions.. there is no such document in 1.2.6 in docs.

On 11/1/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:

Matt wrote:
> Hi,
> I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk
> version.   What do I need to be aware of?  I AM aware 1.2.6 is not the
> newest version, but anything above .6, at this time, seems to have
> stability issues (I've tried them on multiple machines)

/path/to/src/asterisk/docs/UPGRADE.txt or similar file name.

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Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6

[Eric \"ManxPower\" Wieling]

Sorry, the file is located here:

[EMAIL PROTECTED] ~]# ls -l asterisk-1.2.6/UPGRADE.txt
-rw-r--r--  1 1000 1000 8739 Dec  1  2005 asterisk-1.2.6/UPGRADE.txt


Matt wrote:

Thanks for the suggestions.. there is no such document in 1.2.6 in docs.

On 11/1/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:

Matt wrote:
> Hi,
> I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk
> version.   What do I need to be aware of?  I AM aware 1.2.6 is not the
> newest version, but anything above .6, at this time, seems to have
> stability issues (I've tried them on multiple machines)

/path/to/src/asterisk/docs/UPGRADE.txt or similar file name.

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Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6

[Matt]

Ahh yes so it is.   Thanks for the pointer.. seems fairly straight
forward for an upgrade.. guess a test systme is the only way to know
for sure :)

On 11/1/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:

Sorry, the file is located here:

[EMAIL PROTECTED] ~]# ls -l asterisk-1.2.6/UPGRADE.txt
-rw-r--r--  1 1000 1000 8739 Dec  1  2005 asterisk-1.2.6/UPGRADE.txt


Matt wrote:
> Thanks for the suggestions.. there is no such document in 1.2.6 in docs.
>
> On 11/1/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
>> Matt wrote:
>> > Hi,
>> > I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk
>> > version.   What do I need to be aware of?  I AM aware 1.2.6 is not the
>> > newest version, but anything above .6, at this time, seems to have
>> > stability issues (I've tried them on multiple machines)
>>
>> /path/to/src/asterisk/docs/UPGRADE.txt or similar file name.
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[Asterisk-Users] Upgrading Asterisk witk G729 license installed

[Álvaro Palma]

I've an Asterisk 1.2.4 installation, where I've also installed the G729 
codec license. I'd like to upgrade that installation to 1.2.5, but I'm 
not sure if I'll lost the license in the process (and if I'll be able to 
recover it later!!!).


Is there any special consideration I've to keep in mind in this case, or 
should I just run the typical "make + make install" and it will take 
care of keeping the license information?


Thanks a lot for your attention.

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[Asterisk-Users] Upgrading old version of Asteriak - changes

[George Gardiner]

I've spent some time now trying to find information on the changes made 
to the Asterisk config files.  I want to upgrade an old installation to 
the latest version, which of course now uses phone.conf.   If anyone 
could point me in the direction of a set of upgrade notes so that I can 
work out what needs to be changed I would be most grateful.  I'm 
resorting to asking the list as I've not been able to find anything on 
voip-info.org.


Many thanks.

Regards,
George

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Re: [asterisk-users] Upgrading my office - Need help

[Mats Karlsson]

Gary Guthary wrote:

Hi -

If I'm posting this in the wrong place, pease forgive me

Folks, I need help...

One company I consult for is upgrading their office and will need a PBX
replacement in the next two months.

I'm seriously thinking of offering them an 'Asterisk' solution versus them
getting locked in with some PBX vendor.

This means I've got to come up with some sort of demo system to show them.

I've got the hardware. - Dedicated Linux box I can re-config and/or re-load
as needed. - Have one FXS/FXO card to demo intfc to telco. - And a bunch of
Cisco 79xx phones I can use for demos.

If I can get this rolling, I'll be more than happy to pay ***MONEY*** to
anybody who can help. - Also, I'm not afraid to pay for "already-developed"
admin software I can use to manage my system. - I just don't know 'which is
which'.

To prevent cluttering up this board, please send all responses to:
[EMAIL PROTECTED]

Thanks in advance & again apologize if this is not the right place to post
this.

Gary Guthary

Take a look at trixbox.org.

/M

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Re: [asterisk-users] upgrading from 1.4.39 to 1.8.5

[David Backeberg]

read the 1.6 README and the 1.8 README.

If you're using SIP you should expect changes with account
authentication, faxing, output regarding channel status and
performance.

I think that version of 1.4 is late enough you would already be on
DAHDI for hardware devices. If not, you need to convert to DAHDI.

There are also features you might want to try out. Nobody but you
knows how you use your system. Do it on a test system before you put
it into production.

On Fri, Sep 2, 2011 at 8:16 PM, Joseph  wrote:
> What sort of things should I watch out for when upgrading from 1.4.39 to
> 1.8.5
>
> Thanks,
> --
> Joseph
>
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[asterisk-users] upgrading from asterisk 1.4 to 1.6

[p070075 Muhammad Atif Ramzan]

Hi
 I have installed asterisk 1.4 and asterisk-gui 2.0, the problem is that it
cannot upload the .gsm which i record through "voice menu prompt", it gives
error uploading is supported in asterisk 1.6 or higher.
Can anyone help me?


thanks
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Re: [asterisk-users] upgrading from 11.8 to 11.12

[Joseph Kim]

Are there any known complications? I believe this should br straight forward.


IT ENGINEER


 Original message 
From: Olivier 
Date:09/01/2014 10:28 AM (GMT-06:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc:
Subject: [asterisk-users] Asterisk 11.Why two NOTIFY while ringing ?

Hello,

On a Asterisk 11.12.0, I'm studying BLF behaviour with Yealink phones.
My ultimate goal is to present Operator the name and number of every
incoming call so that he/she can if it's worth to pickup a ringing
incoming call.

I've discovered notifycid option in sip.conf.

When a call comes in, I can see that Asterisk is sending two
successive NOTIFY messages while the target is still in Ringing state.

(Unfortunately, Yealink phones display both NOTIFY messages to
Operator but that's another story).

Here is the (anonymized) content of both NOTIFYs:





sip:96@172.16.2.1



sip:96@172.16.2.1


early








sip:+33123456789@172.16.2.1



sip:96@172.16.2.1


early




What is the purpose of these 2 messages ?
Is there a way to get only one ?

Regards

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[asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

[Dmitriy Serov]

Hello.

Several months server working on asterisk 13.7 and pjproject 2.5 
(installed separately). Once a day the server crashes or hangs and is 
familiar sores that written watchdogs.


Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). 
Solved all the problems with compilation I started asterisk several 
times and each time after 5-7 seconds was seg fault.


So I didn't get to use the new version of asterisk. And I really wanted 
to be able to find and remove the cause. I would be grateful for any help.



first segfault:

Program terminated with signal 11, Segmentation fault.
#0  0x7fe19337357e in pj_atomic_dec_and_get (atomic_var=0x1a9a) at 
../src/pj/os_core_unix.c:962

962pj_mutex_lock( atomic_var->mutex );

backtrace: https://ruvoip.net/_other/voip/2016-09-05-1/backtrace-threads.txt

log (16 MB) : https://ruvoip.net/_other/voip/2016-09-05-1/full.txt

second segfault:

Program terminated with signal 11, Segmentation fault.
#0  0x7f5777412788 in pjsip_auth_clt_reinit_req 
(sess=0x7f56f4088a30, rdata=0x7f56ec2b0c98, old_request=0x7f56f47775a8, 
new_request=0x7f56d5b38980) at ../src/pjsip/sip_auth_client.c:1144

1144PJ_ASSERT_RETURN(old_request->msg->type == PJSIP_REQUEST_MSG,

backtrace: https://ruvoip.net/_other/voip/2016-09-05-2/backtrace-threads.txt

log (15 MB) : https://ruvoip.net/_other/voip/2016-09-05-2/full.txt

third segfault:

Program terminated with signal 11, Segmentation fault.
#0  0x7f592bcad53d in pj_pool_alloc (pool=0x7f58002c, size=80) 
at ../include/pj/pool_i.h:60

60void *ptr = pj_pool_alloc_from_block(pool->block_list.next, size);

backtrace: https://ruvoip.net/_other/voip/2016-09-05-3/backtrace-threads.txt

log (11 MB) : https://ruvoip.net/_other/voip/2016-09-05-3/full.txt



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Re: [Asterisk-Users] Upgrading Asterisk witk G729 license installed

[Mojo with Horan & Company, LLC]

I would think that it would be OK to upgrade, but to be sure, your old 
license file should exist at 
/var/lib/asterisk/licenses/G729-.lic and could be backed up from 
there.  After the install, copy this back in.  And make sure you still 
have your codec_g729.so file to put in the modules directory.


Moj

Álvaro Palma wrote:
I've an Asterisk 1.2.4 installation, where I've also installed the G729 
codec license. I'd like to upgrade that installation to 1.2.5, but I'm 
not sure if I'll lost the license in the process (and if I'll be able to 
recover it later!!!).


Is there any special consideration I've to keep in mind in this case, or 
should I just run the typical "make + make install" and it will take 
care of keeping the license information?


Thanks a lot for your attention.



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Re: [Asterisk-Users] Upgrading old version of Asteriak - changes

[Martin Joseph]

On Jun 24, 2006, at 1:37 AM, George Gardiner wrote:

I've spent some time now trying to find information on the changes 
made to the Asterisk config files.  I want to upgrade an old 
installation to the latest version, which of course now uses 
phone.conf.   If anyone could point me in the direction of a set of 
upgrade notes so that I can work out what needs to be changed I would 
be most grateful.  I'm resorting to asking the list as I've not been 
able to find anything on voip-info.org.



Huh,  I never looked at that file before (phone.conf).

Actually they seem to refer to a Linux telephony interface?

Anyone please care to elaborate on what the phone.conf file is really 
for?  The wiki just has a copy of the default file...


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Re: [Asterisk-Users] Upgrading old version of Asteriak - changes

[Brian Capouch]

Martin Joseph wrote:




Huh,  I never looked at that file before (phone.conf).

Actually they seem to refer to a Linux telephony interface?

Anyone please care to elaborate on what the phone.conf file is really 
for?  The wiki just has a copy of the default file...




It's the conf file for the "phone" driver, which also I seem to recall 
is ixj or something.


It interfaces to the "Phone Jack" and "Line Jack" telephony cards.

Google for them.  They've been around forever, IMO they're not very useful.

HTH.

B.

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Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

[George Joseph]

On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov  wrote:

> Hello.
>
> Several months server working on asterisk 13.7 and pjproject 2.5
> (installed separately). Once a day the server crashes or hangs and is
> familiar sores that written watchdogs.
>
> Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5).
> Solved all the problems with compilation I started asterisk several times
> and each time after 5-7 seconds was seg fault.
>
> So I didn't get to use the new version of asterisk. And I really wanted to
> be able to find and remove the cause. I would be grateful for any help.
>
>
All 3 of the backtraces are in different pjproject places which is weird.
Makes me think there's still a library mismatch somewhere.  Is the
separately compiled pjproject still installed on both the build machine and
the server?  They *should* be ignored if --with-pjproject-bundled is
specified but you might want to remove them and try again.



>
> first segfault:
>
> Program terminated with signal 11, Segmentation fault.
> #0  0x7fe19337357e in pj_atomic_dec_and_get (atomic_var=0x1a9a) at
> ../src/pj/os_core_unix.c:962
> 962pj_mutex_lock( atomic_var->mutex );
>
> backtrace: https://ruvoip.net/_other/voip/2016-09-05-1/backtrace-thread
> s.txt
>
> log (16 MB) : https://ruvoip.net/_other/voip/2016-09-05-1/full.txt
>
> second segfault:
>
> Program terminated with signal 11, Segmentation fault.
> #0  0x7f5777412788 in pjsip_auth_clt_reinit_req (sess=0x7f56f4088a30,
> rdata=0x7f56ec2b0c98, old_request=0x7f56f47775a8,
> new_request=0x7f56d5b38980) at ../src/pjsip/sip_auth_client.c:1144
> 1144PJ_ASSERT_RETURN(old_request->msg->type == PJSIP_REQUEST_MSG,
>
> backtrace: https://ruvoip.net/_other/voip/2016-09-05-2/backtrace-thread
> s.txt
>
> log (15 MB) : https://ruvoip.net/_other/voip/2016-09-05-2/full.txt
>
> third segfault:
>
> Program terminated with signal 11, Segmentation fault.
> #0  0x7f592bcad53d in pj_pool_alloc (pool=0x7f58002c, size=80) at
> ../include/pj/pool_i.h:60
> 60void *ptr = pj_pool_alloc_from_block(pool->block_list.next,
> size);
>
> backtrace: https://ruvoip.net/_other/voip/2016-09-05-3/backtrace-thread
> s.txt
>
> log (11 MB) : https://ruvoip.net/_other/voip/2016-09-05-3/full.txt
>
>
>
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>  http://www.asterisk.org/community/astricon-user-conference
>
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Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

[Jonathan H]

All your libraries, kernel, headers and build tools up to date?

The other thing that might be worth noting is the warning along the
lines of "contains modules that were not installed by this version of
Asterisk".

Might be worth deleting anything that appears there, and then starting Asterisk.

On 6 September 2016 at 14:42, George Joseph  wrote:
>
>
> On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov  wrote:
>>
>> Hello.
>>
>> Several months server working on asterisk 13.7 and pjproject 2.5
>> (installed separately). Once a day the server crashes or hangs and is
>> familiar sores that written watchdogs.
>>
>> Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5).
>> Solved all the problems with compilation I started asterisk several times
>> and each time after 5-7 seconds was seg fault.
>>
>> So I didn't get to use the new version of asterisk. And I really wanted to
>> be able to find and remove the cause. I would be grateful for any help.
>>
>
> All 3 of the backtraces are in different pjproject places which is weird.
> Makes me think there's still a library mismatch somewhere.  Is the
> separately compiled pjproject still installed on both the build machine and
> the server?  They should be ignored if --with-pjproject-bundled is specified
> but you might want to remove them and try again.
>
>
>>
>>
>> first segfault:
>>
>> Program terminated with signal 11, Segmentation fault.
>> #0  0x7fe19337357e in pj_atomic_dec_and_get (atomic_var=0x1a9a) at
>> ../src/pj/os_core_unix.c:962
>> 962pj_mutex_lock( atomic_var->mutex );
>>
>> backtrace:
>> https://ruvoip.net/_other/voip/2016-09-05-1/backtrace-threads.txt
>>
>> log (16 MB) : https://ruvoip.net/_other/voip/2016-09-05-1/full.txt
>>
>> second segfault:
>>
>> Program terminated with signal 11, Segmentation fault.
>> #0  0x7f5777412788 in pjsip_auth_clt_reinit_req (sess=0x7f56f4088a30,
>> rdata=0x7f56ec2b0c98, old_request=0x7f56f47775a8,
>> new_request=0x7f56d5b38980) at ../src/pjsip/sip_auth_client.c:1144
>> 1144PJ_ASSERT_RETURN(old_request->msg->type == PJSIP_REQUEST_MSG,
>>
>> backtrace:
>> https://ruvoip.net/_other/voip/2016-09-05-2/backtrace-threads.txt
>>
>> log (15 MB) : https://ruvoip.net/_other/voip/2016-09-05-2/full.txt
>>
>> third segfault:
>>
>> Program terminated with signal 11, Segmentation fault.
>> #0  0x7f592bcad53d in pj_pool_alloc (pool=0x7f58002c, size=80) at
>> ../include/pj/pool_i.h:60
>> 60void *ptr = pj_pool_alloc_from_block(pool->block_list.next,
>> size);
>>
>> backtrace:
>> https://ruvoip.net/_other/voip/2016-09-05-3/backtrace-threads.txt
>>
>> log (11 MB) : https://ruvoip.net/_other/voip/2016-09-05-3/full.txt
>>
>>
>>
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Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

[Dmitriy Serov]

06.09.2016 16:42, George Joseph пишет:



On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov > wrote:


Hello.

Several months server working on asterisk 13.7 and pjproject 2.5
(installed separately). Once a day the server crashes or hangs and
is familiar sores that written watchdogs.

Yesterday I decided to upgrade to 13.11 and bundled pjproject
(2.5.5). Solved all the problems with compilation I started
asterisk several times and each time after 5-7 seconds was seg fault.

So I didn't get to use the new version of asterisk. And I really
wanted to be able to find and remove the cause. I would be
grateful for any help.


All 3 of the backtraces are in different pjproject places which is 
weird.  Makes me think there's still a library mismatch somewhere.  Is 
the separately compiled pjproject still installed on both the build 
machine and the server?  They /should/ be ignored if 
--with-pjproject-bundled is specified but you might want to remove 
them and try again.




George, thank you that you responded. What I was hoping :)
I was also very surprised that SF can happen so quickly and so 
consistently. With all this in completely different places.


When installed separately pjproject source code with bundled not even 
compiled. Of course, I uninstalled and cleaned pjproject according to 
this instructions: 
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject


At the moment I have restore separately pjproject and asterisk 13.7. But 
there is a directory with configured and compiled asterisk 13.11. Happy 
to provide the contents of the autoconfiguration files.
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Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

[George Joseph]

On Tue, Sep 6, 2016 at 7:58 AM, Dmitriy Serov  wrote:

> 06.09.2016 16:42, George Joseph пишет:
>
>
>
> On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov  wrote:
>
>> Hello.
>>
>> Several months server working on asterisk 13.7 and pjproject 2.5
>> (installed separately). Once a day the server crashes or hangs and is
>> familiar sores that written watchdogs.
>>
>> Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5).
>> Solved all the problems with compilation I started asterisk several times
>> and each time after 5-7 seconds was seg fault.
>>
>> So I didn't get to use the new version of asterisk. And I really wanted
>> to be able to find and remove the cause. I would be grateful for any help.
>>
>>
> All 3 of the backtraces are in different pjproject places which is weird.
> Makes me think there's still a library mismatch somewhere.  Is the
> separately compiled pjproject still installed on both the build machine and
> the server?  They *should* be ignored if --with-pjproject-bundled is
> specified but you might want to remove them and try again.
>
>
>
>>
> George, thank you that you responded. What I was hoping :)
> I was also very surprised that SF can happen so quickly and so
> consistently. With all this in completely different places.
>
> When installed separately pjproject source code with bundled not even
> compiled. Of course, I uninstalled and cleaned pjproject according to this
> instructions: https://wiki.asterisk.org/wiki/display/AST/Building+and+
> Installing+pjproject
>
> At the moment I have restore separately pjproject and asterisk 13.7. But
> there is a directory with configured and compiled asterisk 13.11. Happy to
> provide the contents of the autoconfiguration files.
>

config.log and makeopts would be useful if you can zip them up.  Also, what
distribution and version are you running?  As Jonathan suggested, could
there have been any left over or third-party modules left in the asterisk
modules directory?



>
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Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

[Dmitriy Serov]

06.09.2016 16:51, Jonathan H пишет:

All your libraries, kernel, headers and build tools up to date?
I can't be sure because I do not know the required versions of these 
libraries. When you configure and build asterisk, no error detected.
The server has not a new kernel 3.13.6 #4 SMP Wed Aug 20 17:52:10 MSK 
2014 x86_64




The other thing that might be worth noting is the warning along the
lines of "contains modules that were not installed by this version of
Asterisk".

Might be worth deleting anything that appears there, and then starting Asterisk.
I don't use the autoloading of modules. All the necessary modules 
written manually in modules.conf.
At the end of the installation it reported the extra modules (thanks) 
and they removed me as "garbage".



On 6 September 2016 at 14:42, George Joseph  wrote:


On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov  wrote:

Hello.

Several months server working on asterisk 13.7 and pjproject 2.5
(installed separately). Once a day the server crashes or hangs and is
familiar sores that written watchdogs.

Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5).
Solved all the problems with compilation I started asterisk several times
and each time after 5-7 seconds was seg fault.

So I didn't get to use the new version of asterisk. And I really wanted to
be able to find and remove the cause. I would be grateful for any help.


All 3 of the backtraces are in different pjproject places which is weird.
Makes me think there's still a library mismatch somewhere.  Is the
separately compiled pjproject still installed on both the build machine and
the server?  They should be ignored if --with-pjproject-bundled is specified
but you might want to remove them and try again.




first segfault:

Program terminated with signal 11, Segmentation fault.
#0  0x7fe19337357e in pj_atomic_dec_and_get (atomic_var=0x1a9a) at
../src/pj/os_core_unix.c:962
962pj_mutex_lock( atomic_var->mutex );

backtrace:
https://ruvoip.net/_other/voip/2016-09-05-1/backtrace-threads.txt

log (16 MB) : https://ruvoip.net/_other/voip/2016-09-05-1/full.txt

second segfault:

Program terminated with signal 11, Segmentation fault.
#0  0x7f5777412788 in pjsip_auth_clt_reinit_req (sess=0x7f56f4088a30,
rdata=0x7f56ec2b0c98, old_request=0x7f56f47775a8,
new_request=0x7f56d5b38980) at ../src/pjsip/sip_auth_client.c:1144
1144PJ_ASSERT_RETURN(old_request->msg->type == PJSIP_REQUEST_MSG,

backtrace:
https://ruvoip.net/_other/voip/2016-09-05-2/backtrace-threads.txt

log (15 MB) : https://ruvoip.net/_other/voip/2016-09-05-2/full.txt

third segfault:

Program terminated with signal 11, Segmentation fault.
#0  0x7f592bcad53d in pj_pool_alloc (pool=0x7f58002c, size=80) at
../include/pj/pool_i.h:60
60void *ptr = pj_pool_alloc_from_block(pool->block_list.next,
size);

backtrace:
https://ruvoip.net/_other/voip/2016-09-05-3/backtrace-threads.txt

log (11 MB) : https://ruvoip.net/_other/voip/2016-09-05-3/full.txt



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Check us out at: www.digium.com & www.asterisk.org


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Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

[Dmitriy Serov]

06.09.2016 17:08, George Joseph пишет:



On Tue, Sep 6, 2016 at 7:58 AM, Dmitriy Serov > wrote:


06.09.2016 16:42, George Joseph пишет:



On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov
mailto:serov@gmail.com>> wrote:

Hello.

Several months server working on asterisk 13.7 and pjproject
2.5 (installed separately). Once a day the server crashes or
hangs and is familiar sores that written watchdogs.

Yesterday I decided to upgrade to 13.11 and bundled pjproject
(2.5.5). Solved all the problems with compilation I started
asterisk several times and each time after 5-7 seconds was
seg fault.

So I didn't get to use the new version of asterisk. And I
really wanted to be able to find and remove the cause. I
would be grateful for any help.


All 3 of the backtraces are in different pjproject places which
is weird.  Makes me think there's still a library mismatch
somewhere.  Is the separately compiled pjproject still installed
on both the build machine and the server?  They /should/ be
ignored if --with-pjproject-bundled is specified but you might
want to remove them and try again.



George, thank you that you responded. What I was hoping :)
I was also very surprised that SF can happen so quickly and so
consistently. With all this in completely different places.

When installed separately pjproject source code with bundled not
even compiled. Of course, I uninstalled and cleaned pjproject
according to this instructions:
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject



At the moment I have restore separately pjproject and asterisk
13.7. But there is a directory with configured and compiled
asterisk 13.11. Happy to provide the contents of the
autoconfiguration files.


config.log and makeopts would be useful if you can zip them up.  Also, 
what distribution and version are you running?  As Jonathan suggested, 
could there have been any left over or third-party modules left in the 
asterisk modules directory?




https://ruvoip.net/_other/voip/2016-09-05/makeopts_configlogs.zip

Slackware 14.1 x64 with some packages upgraded and some of libraries 
compiled from source.
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Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

[George Joseph]

On Tue, Sep 6, 2016 at 8:40 AM, Dmitriy Serov  wrote:

> 06.09.2016 17:08, George Joseph пишет:
>
>
>
> On Tue, Sep 6, 2016 at 7:58 AM, Dmitriy Serov  wrote:
>
>> 06.09.2016 16:42, George Joseph пишет:
>>
>>
>>
>> On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov 
>> wrote:
>>
>>> Hello.
>>>
>>> Several months server working on asterisk 13.7 and pjproject 2.5
>>> (installed separately). Once a day the server crashes or hangs and is
>>> familiar sores that written watchdogs.
>>>
>>> Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5).
>>> Solved all the problems with compilation I started asterisk several times
>>> and each time after 5-7 seconds was seg fault.
>>>
>>> So I didn't get to use the new version of asterisk. And I really wanted
>>> to be able to find and remove the cause. I would be grateful for any help.
>>>
>>>
>> All 3 of the backtraces are in different pjproject places which is
>> weird.  Makes me think there's still a library mismatch somewhere.  Is the
>> separately compiled pjproject still installed on both the build machine and
>> the server?  They *should* be ignored if --with-pjproject-bundled is
>> specified but you might want to remove them and try again.
>>
>>
>>
>>>
>> George, thank you that you responded. What I was hoping :)
>> I was also very surprised that SF can happen so quickly and so
>> consistently. With all this in completely different places.
>>
>> When installed separately pjproject source code with bundled not even
>> compiled. Of course, I uninstalled and cleaned pjproject according to this
>> instructions: https://wiki.asterisk.org/wiki
>> /display/AST/Building+and+Installing+pjproject
>>
>> At the moment I have restore separately pjproject and asterisk 13.7. But
>> there is a directory with configured and compiled asterisk 13.11. Happy to
>> provide the contents of the autoconfiguration files.
>>
>
> config.log and makeopts would be useful if you can zip them up.  Also,
> what distribution and version are you running?  As Jonathan suggested,
> could there have been any left over or third-party modules left in the
> asterisk modules directory?
>
>
> https://ruvoip.net/_other/voip/2016-09-05/makeopts_configlogs.zip
>


Went through the logs and configs and unfortunately nothing stands out.
The only things I can suggest at this point are to check that there really
are no libpj* files left in /usr/lib, no pj* header files left in
/usr/include, do a system update, make sure that all res_pj* modules are
loaded in modules.conf (maybe try autoload), etc.


>
>
> Slackware 14.1 x64 with some packages upgraded and some of libraries
> compiled from source.
>
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>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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[Asterisk-Users] Upgrading to 1.x from 0.7 on Linux

[Jim Archer]

Hi All...
I have an Asterisk 0.7x server running and have forever now.  I would like 
to upgrade it to 1.0 (or whatever the current version is).  It's running on 
Linux.  I have been told there is now a Debian package for Asterisk on 
Sarge!

I was looking at the Asterisk web site and I noticed that the Wildcard 
X100P cards are deprecated.  I am using two of these cards to interface to 
POTS lines.

If I upgrade to Asterisk 1.x, will I still be able to use these cards?  Are 
there better cards I should look at that will improve quality or offer more 
features?

Thank you very much!
Jim
   
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[asterisk-users] Upgrading to Asterisk 1.4 :: Avoiding the hidden traps

[Johansson Olle E]

In my series of articles about Asterisk 1.4, I've added a checklist  
for those of you upgrading to 1.4 from 1.2.
As always, I appreciate feedback on important things I've forgotten...

http://www.voip-forum.com/category/asterisk/asterisk14/

Have a nice weekend!

/Olle

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[asterisk-users] Upgrading Asterisk and FreePBX from 1.2 to 1.4

[Carlos Chavez]

I have a new customer that wants to upgrade their Asterisk installation
from 1.2.27 to 1.4.22.  They use FreePBX for administration.  Since
there are many syntax and command changes from those versions of
Asterisk, is there an easy way to convert the FreePBX configuration so
it will work with the newer Asterisk?

-- 
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Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to 1.1.1.14

[Scott Keagy]

So I've got phones with ancient firmware, and the release notes for
1.1.1.14 say " read the previous release notes and first upgrade to
1.1.0.16"

 

The 1.1.0.16 firmware is not available for download from the grandstream
website (at least I haven't found it). Any pointers on where to get this
intermediate image? I already tried googling to no avail (didn't help
that I was using a link with 2000 ms latency). Plus, any overall
pointers for making this upgrade process a success would be appreciated.

 

Regards,

Scott

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RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14

[Scott Keagy]

Thanks for your help Claudemir, I look forward to the response. Seems
odd that they don't post an archive of their old firmware versions on
their website, or at least ones that are required to get to the latest
release from whatever is in the field already.

 

Regards,

Scott



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Claudemir
F. Martins
Sent: Saturday, December 02, 2006 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from
1.0.2.13 to1.1.1.14

 

Hi Scott,

I have direct contact with a support person from Grandstream.
I will ask him about that and tell you what did he say as soon as
possible.

Please just wait.

Regards
Claudemir



On 11/30/06, Scott Keagy <[EMAIL PROTECTED]> wrote:

So I've got phones with ancient firmware, and the release notes for
1.1.1.14 say " read the previous release notes and first upgrade to
1.1.0.16"

 

The 1.1.0.16 firmware is not available for download from the grandstream
website (at least I haven't found it). Any pointers on where to get this
intermediate image? I already tried googling to no avail (didn't help
that I was using a link with 2000 ms latency). Plus, any overall
pointers for making this upgrade process a success would be appreciated.

 

Regards,

Scott


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RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14

[Henry.L.Coleman]

Hi  Scott, I have the following firmware
1.1.0.16
1.1.0.11
1.1.1.9
1.1.1.14
1.1.2.6
1.1.2.13

Some of these were not from the official website but they were all an
improvement 1.1.2.13 is very stable apart from the 56 button ext, unit
support.
Let me know which ones you want and I can send them to you.



Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


> Thanks for your help Claudemir, I look forward to the response. Seems
> odd that they don't post an archive of their old firmware versions on
> their website, or at least ones that are required to get to the latest
> release from whatever is in the field already.
>
>
>
> Regards,
>
> Scott
>
> 
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Claudemir
> F. Martins
> Sent: Saturday, December 02, 2006 11:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from
> 1.0.2.13 to1.1.1.14
>
>
>
> Hi Scott,
>
> I have direct contact with a support person from Grandstream.
> I will ask him about that and tell you what did he say as soon as
> possible.
>
> Please just wait.
>
> Regards
> Claudemir
>
>
>
> On 11/30/06, Scott Keagy <[EMAIL PROTECTED]> wrote:
>
> So I've got phones with ancient firmware, and the release notes for
> 1.1.1.14 say " read the previous release notes and first upgrade to
> 1.1.0.16"
>
>
>
> The 1.1.0.16 firmware is not available for download from the grandstream
> website (at least I haven't found it). Any pointers on where to get this
> intermediate image? I already tried googling to no avail (didn't help
> that I was using a link with 2000 ms latency). Plus, any overall
> pointers for making this upgrade process a success would be appreciated.
>
>
>
> Regards,
>
> Scott
>
>
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>
>
>
>
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RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14

[Scott Keagy]

Henry is my newest Hero :)

I'll coordinate with you directly on the releases. Thank you.

Regards,
Scott

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Henry.L.Coleman
Sent: Monday, December 04, 2006 4:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] upgrading grandstream GXP-2000 from
1.0.2.13 to1.1.1.14

Hi  Scott, I have the following firmware
1.1.0.16
1.1.0.11
1.1.1.9
1.1.1.14
1.1.2.6
1.1.2.13

Some of these were not from the official website but they were all an
improvement 1.1.2.13 is very stable apart from the 56 button ext, unit
support.
Let me know which ones you want and I can send them to you.



Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


> Thanks for your help Claudemir, I look forward to the response. Seems
> odd that they don't post an archive of their old firmware versions on
> their website, or at least ones that are required to get to the latest
> release from whatever is in the field already.
>
>
>
> Regards,
>
> Scott
>
> 
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
Claudemir
> F. Martins
> Sent: Saturday, December 02, 2006 11:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from
> 1.0.2.13 to1.1.1.14
>
>
>
> Hi Scott,
>
> I have direct contact with a support person from Grandstream.
> I will ask him about that and tell you what did he say as soon as
> possible.
>
> Please just wait.
>
> Regards
> Claudemir
>
>
>
> On 11/30/06, Scott Keagy <[EMAIL PROTECTED]> wrote:
>
> So I've got phones with ancient firmware, and the release notes for
> 1.1.1.14 say " read the previous release notes and first upgrade to
> 1.1.0.16"
>
>
>
> The 1.1.0.16 firmware is not available for download from the
grandstream
> website (at least I haven't found it). Any pointers on where to get
this
> intermediate image? I already tried googling to no avail (didn't help
> that I was using a link with 2000 ms latency). Plus, any overall
> pointers for making this upgrade process a success would be
appreciated.
>
>
>
> Regards,
>
> Scott
>
>
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Re: [asterisk-users] Upgrading from 1.4 to 11.4.0 - solved (trivial)

[Eric Smith]

Issues below solved by configuring iaxmodem in extensions.conf and in
correcting the context of the initiating telnet script.

... shiny new asterisk

-- 
Eric Smith

Eric Smith wrote on Sun-23-Jun 13  9:24PM
> Hi
> 
> After upgrading from 1.4 to 11.4.0, I am able to receive calls
> and direct them to extensions via defined trunks.
> 
> However, when making outgoing calls I receive the following
> error:
> 
> -- Executing [00044111@default:4] Dial("SIP/fixedline-0004", 
> "SIP/mydevice/0044111,60,w") in new stack
>== Using SIP RTP CoS mark 5
>  -- Called SIP/mydevice/0044111
>  WARNING[13053][C-0002]: channel.c:6164 
> ast_channel_make_compatible_helper: No path to translate from 
> SIP/mydevice-0005 to SIP/hardphone-0004
> 
> And when I try to try to initiate a call with a manager script, I receive an 
> authentication error from the script.
> 
> How might I find more info to help diagnose either or both of these issues?
> 
> -- 
> Eric Smith

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[asterisk-users] Upgrading 13.7 (external pjproject) to 13.9 (bundled pjproject)

[Dmitriy Serov]

Today was another attempt to upgrade to version 13.9 (git).

1. The result was https://issues.asterisk.org/jira/browse/ASTERISK-25970

Had to temporarily block this contact and look forward to advice of how 
to fix it.


2. Also, an unpleasant surprise was the increase in CPU usage from 
10-50% to 200-400% (4 cores). Stable CPU overusage with the same build 
options (DONT_OPTIMIZE, DEBUG_THREADS, BETTER_BACKTRACES, BUILD_NATIVE)


3. After 20 minutes of this work the server has ceased to respond 
promptly to registrations. "core show locks" is attached. Suggestions 
about solving the problem are very welcome.


Dmitriy.


Setting max files open to 30

===
=== 13.9.0-rc1
=== Currently Held Locks
===
===
===   (): 
 (times locked)
===
=== Thread ID: 0x7fbaedb2c700 LWP:25598 (worker_start started at [ 
1077] threadpool.c worker_thread_start())
=== ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields 
&(&object_type->wizards)->lock 0x2243420 (1)
main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D)
main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA)
main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2)
res_pjsip/pjsip_options.c:290 find_an_endpoint()
res_pjsip/pjsip_options.c:342 qualify_contact()
res_pjsip/pjsip_options.c:443 qualify_contact_task()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:1320 execute_tasks()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:351 threadpool_execute()
main/threadpool.c:1103 worker_active()
main/threadpool.c:1024 worker_start()
main/utils.c:1235 dummy_start()
:0 start_thread()
:0 __clone() (0x7fbb8bb68860+6D)
=== ---
===
=== Thread ID: 0x7fbaedab0700 LWP:25596 (worker_start started at [ 
1077] threadpool.c worker_thread_start())
=== ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields 
&(&object_type->wizards)->lock 0x2243420 (1)
main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D)
main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA)
main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2)
res_pjsip/pjsip_options.c:290 find_an_endpoint()
res_pjsip/pjsip_options.c:342 qualify_contact()
res_pjsip/pjsip_options.c:443 qualify_contact_task()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:1320 execute_tasks()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:351 threadpool_execute()
main/threadpool.c:1103 worker_active()
main/threadpool.c:1024 worker_start()
main/utils.c:1235 dummy_start()
:0 start_thread()
:0 __clone() (0x7fbb8bb68860+6D)
=== ---
===
=== Thread ID: 0x7fbb25795700 LWP:25597 (worker_start started at [ 
1077] threadpool.c worker_thread_start())
=== ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields 
&(&object_type->wizards)->lock 0x2243420 (1)
main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D)
main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA)
main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2)
res_pjsip/pjsip_options.c:290 find_an_endpoint()
res_pjsip/pjsip_options.c:342 qualify_contact()
res_pjsip/pjsip_options.c:443 qualify_contact_task()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:1320 execute_tasks()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:351 threadpool_execute()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+8CB8)
main/threadpool.c:1024 worker_start()
main/utils.c:1235 dummy_start()
:0 start_thread()
:0 __clone() (0x7fbb8bb68860+6D)
=== ---
===
=== Thread ID: 0x7fbb254ad700 LWP:25895 (worker_start started at [ 
1077] threadpool.c worker_thread_start())
=== ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields 
&(&object_type->wizards)->lock 0x2243420 (1)
main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D)
main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA)
main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2)
res_pjsip/pjsip_options.c:290 find_an_endpoint()
res_pjsip/pjsip_options.c:342 qualify_contact()
res_pjsip/pjsip_options.c:443 qualify_contact_task()
main/taskprocessor.c:852 ast_taskprocessor_

[asterisk-users] upgrading asterisk 13.13.1 to latest version best practices

[Motty Cruz]

Hello, 

Best practices examples to upgrade Asterisk 13.13.1 to latest version?  

 

Any suggestions? 

 

Thanks,
Motty

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[Asterisk-Users] upgrading to *-1.0.5 on Gentoo; error cdr_mysql.conf': Not found

[Joseph]

I just upgraded from ver. 0.9 to 1.0.5 on Gentoo (from portage) and when
I try to reload my settings I get:

== Unregistered 'mysql' CDR backend
== Parsing '/etc/asterisk/cdr_mysql.conf': Not found (No such file or
directory)

My phone or SPA-3000 will not register with asterisk either.
I've copied the Sip.conf; extenson.conf and iax.conf from old version to
1.0.5

-- 
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Re: [Asterisk-Users] Upgrading to 1.x from 0.7 on Linux

[Walt Reed]

On Fri, May 06, 2005 at 07:43:15PM -0400, Jim Archer said:
> Hi All...
> 
> I have an Asterisk 0.7x server running and have forever now.  I would like 
> to upgrade it to 1.0 (or whatever the current version is).  It's running on 
> Linux.  I have been told there is now a Debian package for Asterisk on 
> Sarge!
> 
> I was looking at the Asterisk web site and I noticed that the Wildcard 
> X100P cards are deprecated.  I am using two of these cards to interface to 
> POTS lines.
> 
> If I upgrade to Asterisk 1.x, will I still be able to use these cards?  Are 
> there better cards I should look at that will improve quality or offer more 
> features?

Yes. The old X100P cards still work fine (in the US, in most cases) with
both 1.x and cvs HEAD (the dev branch.)

That said, I'm migrating a similar setup to one X100P and one SPA3000 to
cut the number of interrupts in half and free up a slot. 


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Re: [Asterisk-Users] Upgrading to 1.x from 0.7 on Linux

[Jim Archer]

Thanks Walt, that's great!

You just remindedme about something, although I don't know why.  When I
first set this up, I wanted Asterisk to detect distinctive ring patterns
and only answer a particular pattern, so that I could share a fax line. 
At the time, it was not possible.  Has this changed?  Will new hardware do
this?

Thanks!

Walt Reed said:
> On Fri, May 06, 2005 at 07:43:15PM -0400, Jim Archer said:
>> Hi All...
>>
>> I have an Asterisk 0.7x server running and have forever now.  I would
>> like
>> to upgrade it to 1.0 (or whatever the current version is).  It's running
>> on
>> Linux.  I have been told there is now a Debian package for Asterisk on
>> Sarge!
>>
>> I was looking at the Asterisk web site and I noticed that the Wildcard
>> X100P cards are deprecated.  I am using two of these cards to interface
>> to
>> POTS lines.
>>
>> If I upgrade to Asterisk 1.x, will I still be able to use these cards?
>> Are
>> there better cards I should look at that will improve quality or offer
>> more
>> features?
>
> Yes. The old X100P cards still work fine (in the US, in most cases) with
> both 1.x and cvs HEAD (the dev branch.)
>
> That said, I'm migrating a similar setup to one X100P and one SPA3000 to
> cut the number of interrupts in half and free up a slot.
>
>
>

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Re: [Asterisk-Users] Upgrading to 1.x from 0.7 on Linux

[Walt Reed]

On Fri, May 06, 2005 at 08:02:43PM -0400, Jim Archer said:
> Thanks Walt, that's great!
> 
> You just remindedme about something, although I don't know why.  When I
> first set this up, I wanted Asterisk to detect distinctive ring patterns
> and only answer a particular pattern, so that I could share a fax line. 
> At the time, it was not possible.  Has this changed?  Will new hardware do
> this?

Yes and no. I had dring setup, but the problem was that I had dring on
both line 1 and line 2, and the code had no way to specify different
contexts on a per channel basis. This has not changed AFAIK. If you only
have dring on one line, and the "fax" number is a dring number (not the
primary number), then I could see it working. Otherwise, probably not.
Dring is really only useful on single ZAP FXO boxes. The SPA-3000 does
not do dring detection on the FXO port IIRC.
 
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Re: [asterisk-users] Upgrading Asterisk and FreePBX from 1.2 to 1.4

[Rob Hillis]

Carlos Chavez wrote:
>   I have a new customer that wants to upgrade their Asterisk installation
> from 1.2.27 to 1.4.22.  They use FreePBX for administration.  Since
> there are many syntax and command changes from those versions of
> Asterisk, is there an easy way to convert the FreePBX configuration so
> it will work with the newer Asterisk?
>   

Unless you have a lot of custom dialplan components in there, the only 
thing you need to be sure of is that you are running FreePBX 2.3 (I 
believe - possibly 2.2) or later.

If you are running a very old version of FreePBX, then you will need to 
upgrade it /before/ you upgrade to Asterisk 1.4.


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Re: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to 1.1.1.14

[Claudemir F. Martins]

Hi Scott,

I have direct contact with a support person from Grandstream.
I will ask him about that and tell you what did he say as soon as possible.

Please just wait.

Regards
Claudemir


On 11/30/06, Scott Keagy <[EMAIL PROTECTED]> wrote:


 So I've got phones with ancient firmware, and the release notes for
1.1.1.14 say " read the previous release notes and first upgrade to
1.1.0.16"



The 1.1.0.16 firmware is not available for download from the grandstream
website (at least I haven't found it). Any pointers on where to get this
intermediate image? I already tried googling to no avail (didn't help that I
was using a link with 2000 ms latency). Plus, any overall pointers for
making this upgrade process a success would be appreciated.



Regards,

Scott

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[asterisk-users] Upgrading to 11.4.0 and ast_channel_make_compatible_helper: No path to translate

[Eric Smith]

Hi

After upgrading from 1.4 to 11.4.0, I *am* able to receive calls
and direct them to extensions via defined trunks.

However, when making outgoing calls I receive the following error:

-- Executing [00044111@default:4] Dial("SIP/fixedline-0004", 
"SIP/mydevice/0044111,60,w") in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/mydevice/0044111
 WARNING[13053][C-0002]: channel.c:6164 ast_channel_make_compatible_helper: 
No path to translate from SIP/mydevice-0005 to SIP/hardphone-0004
  == Spawn extension (default, 0044111, 4) exited non-zero on 
'SIP/fixedline-0004'.
  == MixMonitor close filestream (mixed)

And when I try to try to initiate a call with a manager script, I receive an 
authentication error from the script.

How might I find more info to help diagnose either or both of these issues?

-- 
Eric Smith

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Re: [asterisk-users] Upgrading 13.7 (external pjproject) to 13.9 (bundled pjproject)

[George Joseph]

On Thu, Apr 28, 2016 at 1:10 AM, Dmitriy Serov  wrote:

> Today was another attempt to upgrade to version 13.9 (git).
>
> 1. The result was https://issues.asterisk.org/jira/browse/ASTERISK-25970
>
> Had to temporarily block this contact and look forward to advice of how to
> fix it.
>
> 2. Also, an unpleasant surprise was the increase in CPU usage from 10-50%
> to 200-400% (4 cores). Stable CPU overusage with the same build options
> (DONT_OPTIMIZE, DEBUG_THREADS, BETTER_BACKTRACES, BUILD_NATIVE)
>


Does this also happen with an external pjproject?


>
> 3. After 20 minutes of this work the server has ceased to respond promptly
> to registrations. "core show locks" is attached. Suggestions about solving
> the problem are very welcome.
>
> Dmitriy.
>
>
>
> --
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-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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[asterisk-users] Upgrading from 1.4.21.2 to 1.6.0.5 breaks sql queries with backslashes?

[Kristina Harris]

Hi, all. I've been searching google, bug reports and forums and have 
looked in all the asterisk-users list archives back to 2003 but haven't 
seen an answer to this, so thought I'd post here.

The problem seems to be that Asterisk 1.6.0.5 is sending backslashes 
(needed to escape commas and so forth in 1.4.21.2) as 
*literal* backslashes to Mysql, so that Mysql gives a syntax error and 
many things associated with app_addon_sql_mysql.c fail. I'm pretty sure 
this is an asterisk thing and not an addon thing because the query has the 
backslashes when app_addon_sql_mysql.c gets it whereas I would have 
expected asterisk to have already used them to escape the commas, but I 
could be wrong.

Here's an example of a failing query from extensions.conf (line 2) that 
works fine in 1.4.21.2 and fails in 1.6.0.5:

[vpbx-generic-cc-get-vpbx-number]
exten => s,1,MYSQL(Connect connid ${DBHOST} ${DBUSER} ${DBPW} ${DBNAME})
exten => s,2,MYSQL(Query resultid ${connid} SELECT 
vpbx_id\,vpbx.password\,vpbx.name FROMext\,vpbx WHERE 
mailbox="${CALLERID(num)}" and vpbx_id=vpbx.id)
exten => s,3,MYSQL(Fetch fetchid ${resultid} vpbx-id vpbx-password vpbx)
exten => s,4,MYSQL(Clear ${resultid})
exten => s,5,MYSQL(Query resultid ${connid} SELECT mailbox FROM ext\,devtype 
WHERE active="Y" and ext.devtype_id=devtype.id and devtype.model like 
"AUTOATTENDANT%" and vpbx_id=${vpbx-id})
exten => s,6,MYSQL(Fetch fetchid ${resultid} mailbox)
exten => s,7,MYSQL(Clear ${resultid})
exten => s,8,Set(vpbx-number=${mailbox})
exten => s,9,MYSQL(Disconnect ${connid})
exten => s,10,Return

And here's the error from 1.6.0.5:

[Apr 22 09:06:36] WARNING[17379]: app_addon_sql_mysql.c:311 aMYSQL_query: 
aMYSQL_query: mysql_query failed. Error: You have an error in your SQL 
syntax; check the manual that corresponds to your MySQL server version for 
the right syntax to use near '\,vpbx.password\,vpbx.name FROM ext\,vpbx 
WHERE mailbox="1234567890" and vpbx_id' at line 1

If I remove *all* backslashes in the Query line, it works fine in 1.6.0.5. 
If I add backslashes to all spaces and double-quotes, it fails in 1.6.0.5.

I have added all three "[compat]" options I could find to asterisk.conf 
although none of them would seem to affect this problem:

[compat]
app_set=1.4
pbx_realtime=1.4
res_agi=1.4

I have looked through all the UPGRADE instructions, but haven't found 
anything helpful.

It seems so odd that suddenly backslashes would not only not be required
but that they wouldn't work at all. Sure, I could just remove all the
backslashes in the configs for 1.6 and it would work, but it would be a
much smoother upgrade (and smoother fallback, if necessary) if the config
file changes are as minimal as possible between versions.

Anyone have any ideas? Am I just missing some compatibility option (or
combination thereof) somewhere?

Thanks for any info!

Kristina

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