Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

[Jöran Vinzens]

Hi,

Using SIPp to check your asterisk is working has some pitfalls.
We recorded SIP invites (as these are the important parts of a call) of a
normal call going through the Asterisk. We recorded it the old working way.
In General we just compared what we had before with what we have after the
upgrade. So if the Headers are Correct (from, to, our own X header, P
Header etc). Since there are a couple of things you might want to check,
e.g. someone is not allowed to place long distance calls, you can check the
behavior of the asterisk as well so if the calls get rejected when some
user dials some number.

In SIPp there are these Scenario Files (XML Files) that contain a sequence
of SIP Messages to send/receive. Using the receiving of Messages you can
specifically check for presence or absence of a Header or a field in a
header.

there are lots of examples in the github repo
https://github.com/SIPp/sipp

For a A calls B call you need to start two SIPp Instances (one sending the
call, one receiving the call)
If your clients register to your Asterisk no not forget to do so, otherwise
the Asterisk has no AOR to forward the call to. (Using plain UDP helps here
a lot).

The first check you build up might be some more work even if you never
played around with SIPp but all what follows are quite simple and ensure
quality.

for my talk i put everything together you might need to place a simple call
to an asterisk
https://github.com/sipgate/signaling-test

the start shell script will start first a registration to your Asterisk and
then starts two sipp instances to place the call.

feel free to use it.

BR
Jöran

On Mon, Dec 7, 2020 at 8:29 PM Eric Wieling  wrote:

> I'm sure you can, but I've never done it.
>
> On 12/7/20 2:18 PM, the...@sys-concept.com wrote:
> > Sound reasonable.  I know it take time to debug the dial-plan after
> upgrade.
> >
> > Can I use sipp, from command line to call my local asterisk specific
> > extension and to observe in another terminal via "asterisk -vvr"
> > what it is doing?
> >
> >
> > On 12/07/2020 11:50 AM, Eric Wieling wrote:
> >> Read UPGRADE.TXT in v13 and v16.  Then read it again.
> >>
> >> I upgraded from Asterisk v11 to Asterisk v13.  Once all issues were
> >> resolved, then I switched to PJSIP.   Once all the issues with PJSIP
> >> were resolved, then I upgraded from v13 to Asterisk v16.   This was done
> >> over the course of about a year, but I was not in any hurry.
> >>
> >> PJSIP configuration is fundamentally different chan_sip configuration. I
> >> don't recommend switching to PJSIP and upgrade Asterisk at the same
> time.
> >>
> >> On 12/6/20 3:38 PM, the...@sys-concept.com wrote:
> >>> I'm planning to upgrade my asterisk-11.25 to ver. 13
> >>> or should I go to 11 to 16
> >>>
> >>> Is there any official documentation how to upgrade, what to watch for
> >>> during upgrade?
> >>>
> >>>
> >>
> >
>
> --
> http://help.nyigc.net/
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
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-- 

Jöran Vinzens - vinz...@sipgate.de
Telefon: +49 211-63 55 56-21
Telefax: +49 211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391

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Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

[Eric Wieling]

I'm sure you can, but I've never done it.

On 12/7/20 2:18 PM, the...@sys-concept.com wrote:

Sound reasonable.  I know it take time to debug the dial-plan after upgrade.

Can I use sipp, from command line to call my local asterisk specific
extension and to observe in another terminal via "asterisk -vvr"
what it is doing?


On 12/07/2020 11:50 AM, Eric Wieling wrote:

Read UPGRADE.TXT in v13 and v16.  Then read it again.

I upgraded from Asterisk v11 to Asterisk v13.  Once all issues were
resolved, then I switched to PJSIP.   Once all the issues with PJSIP
were resolved, then I upgraded from v13 to Asterisk v16.   This was done
over the course of about a year, but I was not in any hurry.

PJSIP configuration is fundamentally different chan_sip configuration. I
don't recommend switching to PJSIP and upgrade Asterisk at the same time.

On 12/6/20 3:38 PM, the...@sys-concept.com wrote:

I'm planning to upgrade my asterisk-11.25 to ver. 13
or should I go to 11 to 16

Is there any official documentation how to upgrade, what to watch for
during upgrade?








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Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

[thelma]

Sound reasonable.  I know it take time to debug the dial-plan after upgrade.

Can I use sipp, from command line to call my local asterisk specific
extension and to observe in another terminal via "asterisk -vvr"
what it is doing?


On 12/07/2020 11:50 AM, Eric Wieling wrote:
> Read UPGRADE.TXT in v13 and v16.  Then read it again.
> 
> I upgraded from Asterisk v11 to Asterisk v13.  Once all issues were
> resolved, then I switched to PJSIP.   Once all the issues with PJSIP
> were resolved, then I upgraded from v13 to Asterisk v16.   This was done
> over the course of about a year, but I was not in any hurry.
> 
> PJSIP configuration is fundamentally different chan_sip configuration. I
> don't recommend switching to PJSIP and upgrade Asterisk at the same time.
> 
> On 12/6/20 3:38 PM, the...@sys-concept.com wrote:
>> I'm planning to upgrade my asterisk-11.25 to ver. 13
>> or should I go to 11 to 16
>>
>> Is there any official documentation how to upgrade, what to watch for
>> during upgrade?
>>
>>
> 

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Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

[Eric Wieling]

Read UPGRADE.TXT in v13 and v16.  Then read it again.

I upgraded from Asterisk v11 to Asterisk v13.  Once all issues were 
resolved, then I switched to PJSIP.   Once all the issues with PJSIP 
were resolved, then I upgraded from v13 to Asterisk v16.   This was done 
over the course of about a year, but I was not in any hurry.


PJSIP configuration is fundamentally different chan_sip configuration. 
I don't recommend switching to PJSIP and upgrade Asterisk at the same 
time.


On 12/6/20 3:38 PM, the...@sys-concept.com wrote:

I'm planning to upgrade my asterisk-11.25 to ver. 13
or should I go to 11 to 16

Is there any official documentation how to upgrade, what to watch for
during upgrade?




--
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Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

[thelma]

On 12/07/2020 05:06 AM, Jöran Vinzens wrote:
> Hi,
> 
> I guess describing how SIPp works here on a mailliste might be too much.
> But if you do not want to prove your setup automatically, you do not need
> to know SIPp.
> 
> But there was a talk in 2014 Astricon giving an overview about SIP Testing
> with SIPp
> https://www.youtube.com/watch?v=TZMrPJM4HMc
> 
> BR
> Jöran
> 
> 
> On Sun, Dec 6, 2020 at 11:25 PM  wrote:
> 
>> On 12/06/2020 01:44 PM, Jöran Vinzens wrote:
>>> Hi,
>>>
>>> I did a talk on Astricon 2019 on this topic. Unfortunately there are no
>>> videos of that year but you can find my slides here covering some
>> pitfalls.
>>>
>> https://www.slideshare.net/JranVinzens/asterisk-11to16-what-could-go-wrong
>>>
>>> Good luck by updating.
>>>
>>> BR
>>> Jöran
>>>
>>>
>>>  schrieb am So., 6. Dez. 2020, 21:40:
>>>
 I'm planning to upgrade my asterisk-11.25 to ver. 13
 or should I go to 11 to 16

 Is there any official documentation how to upgrade, what to watch for
 during upgrade?

>> Thanks for the input. I've never run SIPp signaling test.
>> Is there more information how to implement it?

Thanks, yes initially it looked interesting.  But I don't see how "sipp"
can be use to test my extension.conf dial plan.


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Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

[Jöran Vinzens]

Hi,

I guess describing how SIPp works here on a mailliste might be too much.
But if you do not want to prove your setup automatically, you do not need
to know SIPp.

But there was a talk in 2014 Astricon giving an overview about SIP Testing
with SIPp
https://www.youtube.com/watch?v=TZMrPJM4HMc

BR
Jöran


On Sun, Dec 6, 2020 at 11:25 PM  wrote:

> On 12/06/2020 01:44 PM, Jöran Vinzens wrote:
> > Hi,
> >
> > I did a talk on Astricon 2019 on this topic. Unfortunately there are no
> > videos of that year but you can find my slides here covering some
> pitfalls.
> >
> https://www.slideshare.net/JranVinzens/asterisk-11to16-what-could-go-wrong
> >
> > Good luck by updating.
> >
> > BR
> > Jöran
> >
> >
> >  schrieb am So., 6. Dez. 2020, 21:40:
> >
> >> I'm planning to upgrade my asterisk-11.25 to ver. 13
> >> or should I go to 11 to 16
> >>
> >> Is there any official documentation how to upgrade, what to watch for
> >> during upgrade?
> >>
> Thanks for the input. I've never run SIPp signaling test.
> Is there more information how to implement it?
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> 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



-- 

Jöran Vinzens - vinz...@sipgate.de
Telefon: +49 211-63 55 56-21
Telefax: +49 211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391

www.sipgate.de - www.sipgate.co.uk
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Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

[thelma]

On 12/06/2020 01:44 PM, Jöran Vinzens wrote:
> Hi,
> 
> I did a talk on Astricon 2019 on this topic. Unfortunately there are no
> videos of that year but you can find my slides here covering some pitfalls.
> https://www.slideshare.net/JranVinzens/asterisk-11to16-what-could-go-wrong
> 
> Good luck by updating.
> 
> BR
> Jöran
> 
> 
>  schrieb am So., 6. Dez. 2020, 21:40:
> 
>> I'm planning to upgrade my asterisk-11.25 to ver. 13
>> or should I go to 11 to 16
>>
>> Is there any official documentation how to upgrade, what to watch for
>> during upgrade?
>>
Thanks for the input. I've never run SIPp signaling test.
Is there more information how to implement it?

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Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

[Jöran Vinzens]

Hi,

I did a talk on Astricon 2019 on this topic. Unfortunately there are no
videos of that year but you can find my slides here covering some pitfalls.
https://www.slideshare.net/JranVinzens/asterisk-11to16-what-could-go-wrong

Good luck by updating.

BR
Jöran


 schrieb am So., 6. Dez. 2020, 21:40:

> I'm planning to upgrade my asterisk-11.25 to ver. 13
> or should I go to 11 to 16
>
> Is there any official documentation how to upgrade, what to watch for
> during upgrade?
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> 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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[asterisk-users] upgrade asterisk 11 to 13 or 11-16

[thelma]

I'm planning to upgrade my asterisk-11.25 to ver. 13
or should I go to 11 to 16

Is there any official documentation how to upgrade, what to watch for
during upgrade?


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[asterisk-users] Upgrade to Fedora 21, now gv requires rtp ?

[sean darcy]

I just upgraded to fedora 21. I'm running asterisk 11.6.0. All works 
with Fedora 20.



-- Executing [s@DialOut:15] Dial(DAHDI/1-1, 
motif/8447/+1212xxxy...@voice.google.com,,rTt) in new stack
[Mar  1 21:24:06] ERROR[2477][C-]: rtp_engine.c:259 
ast_rtp_instance_new: No RTP engine was found. Do you have one loaded?
[Mar  1 21:24:06] ERROR[2477][C-]: chan_motif.c:1820 
jingle_request: Unable to create Jingle session on endpoint '8447'




any help appreciated.

sean


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Re: [asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

[Mike Diehl]

Thank you!  That was very helpful.

Mike.

On Wed, Jul 10, 2013 at 7:38 PM, Matthew Jordan mjor...@digium.com wrote:

 On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl mdiehlena...@gmail.com wrote:

 Hi all,

 I'm contemplating an upgrade from 10.2.4 to 11.4.x.  However, the
 1.8.x to 10.4.x upgrade was painful; some of the modules had been
 renamed, if I recall correctly.

 So, is there a list of MAJOR changes and GOTCHA's between 10.x and
 11.x?  I'm hoping for something a little less granular than the
 release notes from 10.2.x to 11.4.x.  I don't mind reading, but that
 is almost as long as War and Peace!

 Does such a document exist, or do I need to start reading..


 Upgrade notes:

 https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

 While the upgrade notes cover changes to configuration and module status, it
 is also a good idea to read through what is new:

 https://wiki.asterisk.org/wiki/display/AST/New+in+11

 I wouldn't say it is War and Peace, but yes, there is some content in
 there.

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

[Eric Wieling]

Similar information is included in every Asterisk source tarball as UPGRADE*.txt

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Thursday, July 11, 2013 3:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's 
Digest version?

Thank you!  That was very helpful.

Mike.

On Wed, Jul 10, 2013 at 7:38 PM, Matthew Jordan mjor...@digium.com wrote:

 On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl mdiehlena...@gmail.com wrote:

 Hi all,

 I'm contemplating an upgrade from 10.2.4 to 11.4.x.  However, the 
 1.8.x to 10.4.x upgrade was painful; some of the modules had been 
 renamed, if I recall correctly.

 So, is there a list of MAJOR changes and GOTCHA's between 10.x and 
 11.x?  I'm hoping for something a little less granular than the 
 release notes from 10.2.x to 11.4.x.  I don't mind reading, but that 
 is almost as long as War and Peace!

 Does such a document exist, or do I need to start reading..


 Upgrade notes:

 https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

 While the upgrade notes cover changes to configuration and module 
 status, it is also a good idea to read through what is new:

 https://wiki.asterisk.org/wiki/display/AST/New+in+11

 I wouldn't say it is War and Peace, but yes, there is some content 
 in there.

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: 
 http://digium.com  http://asterisk.org

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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[asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

[Mike Diehl]

Hi all,

I'm contemplating an upgrade from 10.2.4 to 11.4.x.  However, the
1.8.x to 10.4.x upgrade was painful; some of the modules had been
renamed, if I recall correctly.

So, is there a list of MAJOR changes and GOTCHA's between 10.x and
11.x?  I'm hoping for something a little less granular than the
release notes from 10.2.x to 11.4.x.  I don't mind reading, but that
is almost as long as War and Peace!

Does such a document exist, or do I need to start reading..

TIA,

Mike Diehl.

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Re: [asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

[Matthew Jordan]

On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl mdiehlena...@gmail.com wrote:

 Hi all,

 I'm contemplating an upgrade from 10.2.4 to 11.4.x.  However, the
 1.8.x to 10.4.x upgrade was painful; some of the modules had been
 renamed, if I recall correctly.

 So, is there a list of MAJOR changes and GOTCHA's between 10.x and
 11.x?  I'm hoping for something a little less granular than the
 release notes from 10.2.x to 11.4.x.  I don't mind reading, but that
 is almost as long as War and Peace!

 Does such a document exist, or do I need to start reading..


Upgrade notes:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

While the upgrade notes cover changes to configuration and module status,
it is also a good idea to read through what is new:

https://wiki.asterisk.org/wiki/display/AST/New+in+11

I wouldn't say it is War and Peace, but yes, there is some content in
there.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org
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Re: [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0

[Dennis Dryden]

AsteriskNOW 1 and 2 are both based on Centos 5 and I think the new
AsteriskNOW 3 is based on Centos 6 so upgrades are not supported by the
Linux OS distribution[1]. It's best to backup and reinstall with the new
version. It's a shame AsteriskNOW is not based on Debian so it could be
dist-upgraded between versions.

Cheers,
Dennis


[1] http://wiki.centos.org/HowTos/MigrationGuide/MigratingFiveToSix


On Mon, May 13, 2013 at 10:27 PM, Andre Goree an...@drenet.net wrote:

 Hello all.  I was hoping someone out there might have some advice or
 suggestions regarding an upgrade from an archaic Asterisk version.

 I've been given the daunting task of upgrading a very old Asterisk-1.0.x
 install to a recent LTS version.  I'll also need the install to have
 high-availability and failover support.

 From my research, it would appear that AsteriskNOW-3.0 might be my best
 bet, as it seems to be running Asterisk-11.  I've previously installed
 Asterisk-11+FreePBX in a VM, and this appears to be very similar.  Is there
 any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than the
 obvious fact that everything is nicely placed on an iso for ease of
 installation?

 As for the actual upgrade, is it possible to step through each of the
 UPGRADE*.txt files under the Asterisk-11 source?  I.e, UPGRADE-1.2.txt -
 UPGRADE-1.4.txt - UPGRADE-1.6.txt - UPGRADE-1.8.txt - UPGRADE.txt? Or
 would it be prudent to recreate my current 1.0.x configuration under
 Asterisk-11 instead?

 Regarding HA/failover, is a hardware solution (such as Digium's R800/R850)
 my only option?  During my research I've found scripts on the internet that
 allow for failover using arp/nmap, however those appear to only work for
 hardware failures.  I would need something that can account for both
 hardware and software failures.

 Thanks in advance for any advice that anyone can give on the subject.  Any
 suggestions, etc. would help immensely!

 --
 Andre Goree
 -=-=-=-=-=-
 Email - an...@drenet.net
 Website   - http://blog.drenet.net
 PGP key   - 
 http://www.drenet.net/**0x83ADAAAB.aschttp://www.drenet.net/0x83ADAAAB.asc
 -=-=-=-=-=-

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Re: [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0

[Andre Goree]

On 2013-05-14 3:50 am, Dennis Dryden wrote:

AsteriskNOW 1 and 2 are both based on Centos 5 and I think the new
AsteriskNOW 3 is based on Centos 6 so upgrades are not supported by
the Linux OS distribution[1]. It's best to backup and reinstall with
the new version. It's a shame AsteriskNOW is not based on Debian so it
could be dist-upgraded between versions.

Cheers,
Dennis

[1] http://wiki.centos.org/HowTos/MigrationGuide/MigratingFiveToSix [6]




Thanks for the reply!  Keep in mind, I'm wanting to go from vanilla 
Asterisk 1.0.x to AsteriskNOW 3 -- or Asterisk 11 + FreePBX.  This will 
be on all new hardware -- which I guess actually answers my question 
regarding the feasibility of an in-place upgrade.  From what I can tell, 
my only choice will be to rebuild the configuration that is currently in 
the 1.0.x install using AsteriskNOW/Asterisk+FreePBX.  Probably won't be 
fun :/


Any insight on failover solutions?



On Mon, May 13, 2013 at 10:27 PM, Andre Goree an...@drenet.net wrote:

Hello all.  I was hoping someone out there might have some advice or 
suggestions regarding an upgrade from an archaic Asterisk version.


I've been given the daunting task of upgrading a very old 
Asterisk-1.0.x install to a recent LTS version.  I'll also need the 
install to have high-availability and failover support.


From my research, it would appear that AsteriskNOW-3.0 might be my best 
bet, as it seems to be running Asterisk-11.  I've previously installed 
Asterisk-11+FreePBX in a VM, and this appears to be very similar.  Is 
there any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than 
the obvious fact that everything is nicely placed on an iso for ease of 
installation?


As for the actual upgrade, is it possible to step through each of the 
UPGRADE*.txt files under the Asterisk-11 source?  I.e, UPGRADE-1.2.txt 
- UPGRADE-1.4.txt - UPGRADE-1.6.txt - UPGRADE-1.8.txt - 
UPGRADE.txt? Or would it be prudent to recreate my current 1.0.x 
configuration under Asterisk-11 instead?


Regarding HA/failover, is a hardware solution (such as Digium's 
R800/R850) my only option?  During my research I've found scripts on 
the internet that allow for failover using arp/nmap, however those 
appear to only work for hardware failures.  I would need something that 
can account for both hardware and software failures.


Thanks in advance for any advice that anyone can give on the subject. 
 Any suggestions, etc. would help immensely!


--
Andre Goree
-=-=-=-=-=-
Email     - an...@drenet.net
Website   - http://blog.drenet.net [1]
PGP key   - http://www.drenet.net/0x83ADAAAB.asc [2]
-=-=-=-=-=-

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Links:
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[2] http://www.drenet.net/0x83ADAAAB.asc
[3] http://www.api-digital.com
[4] http://www.asterisk.org/hello
[5] http://lists.digium.com/mailman/listinfo/asterisk-users
[6] http://wiki.centos.org/HowTos/MigrationGuide/MigratingFiveToSix

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Re: [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0

[Kevin Larsen]

I can't comment on an Asterisk to Asterisk migration, but I have done a 
migration from a different pbx to Asterisk where no downtime was allowed. 
What we did was put the new instance (Asterisk) as the primary call 
handler with a rule that said anythnig that isn't a match gets sent over 
to the secondary PBX (in your case it would be the Asterisk 1.0.x 
instance). Then you will also have to make corresponding changes on the 
original PBX to send the appropriate calls/extensions back to the new 
Asterisk instance. In our case, we started with Asterisk as a voicemail 
box, then moved one department at a time over so we could do testing to 
make sure it was working. By the end, we were down to all the extensions 
that didn't have any kind of special handling on them and we moved those 
over as we had time/money for new phones.

Planning, having test extensions and moving slowly at the start are your 
friend. Once you understand all the ins and outs of the migration, you can 
start moving to the new instance on a faster pace. It is possible to do it 
with virtually no downtime.

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From:   Andre Goree an...@drenet.net
To: asterisk-users@lists.digium.com, 
Date:   05/14/2013 08:24 AM
Subject:Re: [asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0
Sent by:asterisk-users-boun...@lists.digium.com



On 2013-05-14 3:50 am, Dennis Dryden wrote:
 AsteriskNOW 1 and 2 are both based on Centos 5 and I think the new
 AsteriskNOW 3 is based on Centos 6 so upgrades are not supported by
 the Linux OS distribution[1]. It's best to backup and reinstall with
 the new version. It's a shame AsteriskNOW is not based on Debian so it
 could be dist-upgraded between versions.
 
 Cheers,
 Dennis
 
 [1] http://wiki.centos.org/HowTos/MigrationGuide/MigratingFiveToSix [6]
 


Thanks for the reply!  Keep in mind, I'm wanting to go from vanilla 
Asterisk 1.0.x to AsteriskNOW 3 -- or Asterisk 11 + FreePBX.  This will 
be on all new hardware -- which I guess actually answers my question 
regarding the feasibility of an in-place upgrade.  From what I can tell, 
my only choice will be to rebuild the configuration that is currently in 
the 1.0.x install using AsteriskNOW/Asterisk+FreePBX.  Probably won't be 
fun :/

Any insight on failover solutions?


 On Mon, May 13, 2013 at 10:27 PM, Andre Goree an...@drenet.net wrote:
 
 Hello all.  I was hoping someone out there might have some advice or 
 suggestions regarding an upgrade from an archaic Asterisk version.
 
 I've been given the daunting task of upgrading a very old 
 Asterisk-1.0.x install to a recent LTS version.  I'll also need the 
 install to have high-availability and failover support.
 
 From my research, it would appear that AsteriskNOW-3.0 might be my best 
 bet, as it seems to be running Asterisk-11.  I've previously installed 
 Asterisk-11+FreePBX in a VM, and this appears to be very similar.  Is 
 there any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than 
 the obvious fact that everything is nicely placed on an iso for ease of 
 installation?
 
 As for the actual upgrade, is it possible to step through each of the 
 UPGRADE*.txt files under the Asterisk-11 source?  I.e, UPGRADE-1.2.txt 
 - UPGRADE-1.4.txt - UPGRADE-1.6.txt - UPGRADE-1.8.txt - 
 UPGRADE.txt? Or would it be prudent to recreate my current 1.0.x 
 configuration under Asterisk-11 instead?
 
 Regarding HA/failover, is a hardware solution (such as Digium's 
 R800/R850) my only option?  During my research I've found scripts on 
 the internet that allow for failover using arp/nmap, however those 
 appear to only work for hardware failures.  I would need something that 
 can account for both hardware and software failures.
 
 Thanks in advance for any advice that anyone can give on the subject. 
  Any suggestions, etc. would help immensely!
 
 --
 Andre Goree
 -=-=-=-=-=-
 Email - an...@drenet.net
 Website   - http://blog.drenet.net [1]
 PGP key   - http://www.drenet.net/0x83ADAAAB.asc [2]
 -=-=-=-=-=-
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com [3] 
 --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello [4]
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users [5]
 
 
 
 Links:
 --
 [1] http://blog.drenet.net
 [2] http://www.drenet.net/0x83ADAAAB.asc
 [3] http://www.api-digital.com
 [4] http://www.asterisk.org/hello
 [5] http://lists.digium.com/mailman/listinfo/asterisk-users
 [6] http://wiki.centos.org/HowTos/MigrationGuide/MigratingFiveToSix
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
 http

[asterisk-users] Upgrade from 1.0.x to AsteriskNOW 3.0

[Andre Goree]

Hello all.  I was hoping someone out there might have some advice or 
suggestions regarding an upgrade from an archaic Asterisk version.


I've been given the daunting task of upgrading a very old Asterisk-1.0.x 
install to a recent LTS version.  I'll also need the install to have 
high-availability and failover support.


From my research, it would appear that AsteriskNOW-3.0 might be my best 
bet, as it seems to be running Asterisk-11.  I've previously installed 
Asterisk-11+FreePBX in a VM, and this appears to be very similar.  Is 
there any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than 
the obvious fact that everything is nicely placed on an iso for ease of 
installation?


As for the actual upgrade, is it possible to step through each of the 
UPGRADE*.txt files under the Asterisk-11 source?  I.e, UPGRADE-1.2.txt 
- UPGRADE-1.4.txt - UPGRADE-1.6.txt - UPGRADE-1.8.txt - UPGRADE.txt? 
Or would it be prudent to recreate my current 1.0.x configuration under 
Asterisk-11 instead?


Regarding HA/failover, is a hardware solution (such as Digium's 
R800/R850) my only option?  During my research I've found scripts on the 
internet that allow for failover using arp/nmap, however those appear to 
only work for hardware failures.  I would need something that can 
account for both hardware and software failures.


Thanks in advance for any advice that anyone can give on the subject.  
Any suggestions, etc. would help immensely!


--
Andre Goree
-=-=-=-=-=-
Email - an...@drenet.net
Website   - http://blog.drenet.net
PGP key   - http://www.drenet.net/0x83ADAAAB.asc
-=-=-=-=-=-

--
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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Administrator TOOTAI]

Le 02/06/2012 19:18, Administrator TOOTAI a écrit :

Le 30/05/2012 15:02, Andres a écrit :




Considering that you made progress on your initial problem by setting
nat=force_rport (resulting in connected calls with no audio) and now
you're mentioning the use of externaddr, I'd recommend a very
careful reading of the NAT SUPPORT section of sip.conf.sample in the
configs directory of the Asterisk source tree.  In Asterisk 1.8, there
is a new configuration option named media_address which may be of
particular interest.
It sounds like a NAT issue to me too.  Why don't you do a quick test 
and put the Asterisk box on a public IP if you can.  If it works, you 
will have narrowed down the issue to a NAT problem.   You could have 
a nat router with a broken SIP ALG.




Back to the story: even out of VM -which means on a public IP- the 
timeout problem till appears. And more odd, if a communication start, 
the call get hanged up because of this timeout :-(


All peers and users are setted with nat=yes, phones connected to 
Asterisk have directmedia=nonat and peers gateways have directmedia=yes.


Remember, we only face this problem with Dellmont services and 
asterisk 1.8/10. Previous asterisk versions are working well.


Does someone else use Dellmont services (VoipBuster, SipDiscount, 
Intenetcalls, Voicetrading, ...) with asterisk 1.8 or 10? If yes and 
without problem, would it be possible to share configurations?


Thanks for your help.



For the archives.

Problem was with Dellmont services: no audio or calls stopping after 120 
seconds. They gave me another IP for setting outgoing calls and now 
everything is going smoothly with both versions.


Thanks for help.

--
Daniel

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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Administrator TOOTAI]

Le 30/05/2012 15:02, Andres a écrit :




Considering that you made progress on your initial problem by setting
nat=force_rport (resulting in connected calls with no audio) and now
you're mentioning the use of externaddr, I'd recommend a very
careful reading of the NAT SUPPORT section of sip.conf.sample in the
configs directory of the Asterisk source tree.  In Asterisk 1.8, there
is a new configuration option named media_address which may be of
particular interest.
It sounds like a NAT issue to me too.  Why don't you do a quick test 
and put the Asterisk box on a public IP if you can.  If it works, you 
will have narrowed down the issue to a NAT problem.   You could have a 
nat router with a broken SIP ALG.




Back to the story: even out of VM -which means on a public IP- the 
timeout problem till appears. And more odd, if a communication start, 
the call get hanged up because of this timeout :-(


All peers and users are setted with nat=yes, phones connected to 
Asterisk have directmedia=nonat and peers gateways have directmedia=yes.


Remember, we only face this problem with Dellmont services and asterisk 
1.8/10. Previous asterisk versions are working well.


Does someone else use Dellmont services (VoipBuster, SipDiscount, 
Intenetcalls, Voicetrading, ...) with asterisk 1.8 or 10? If yes and 
without problem, would it be possible to share configurations?


Thanks for your help.

--
Daniel

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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Matthew J. Roth]

Administrator TOOTAI wrote:

 10.0.70.12 is the IP of Asterisk server (kvm virtual machine) which is 
 replaced by externaddr parameter from sip.conf.
 
 If you have other ideas, welcome ;-)


Considering that you made progress on your initial problem by setting
nat=force_rport (resulting in connected calls with no audio) and now
you're mentioning the use of externaddr, I'd recommend a very
careful reading of the NAT SUPPORT section of sip.conf.sample in the
configs directory of the Asterisk source tree.  In Asterisk 1.8, there
is a new configuration option named media_address which may be of
particular interest.

This is confusing because your first email said you had nat=no in
your working 1.6.24 setup, but everything you're saying indicates a
NAT problem to me.  A diagram showing all network elements between
your Asterisk server and the remote host would be helpful.  To avoid
further confusion, please include full and unaltered logs, SIP traces,
and configurations in future posts.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Andres]

Considering that you made progress on your initial problem by setting
nat=force_rport (resulting in connected calls with no audio) and now
you're mentioning the use of externaddr, I'd recommend a very
careful reading of the NAT SUPPORT section of sip.conf.sample in the
configs directory of the Asterisk source tree.  In Asterisk 1.8, there
is a new configuration option named media_address which may be of
particular interest.
   
It sounds like a NAT issue to me too.  Why don't you do a quick test and 
put the Asterisk box on a public IP if you can.  If it works, you will 
have narrowed down the issue to a NAT problem.   You could have a nat 
router with a broken SIP ALG.


--
Technical Support
http://www.cellroute.net


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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Administrator TOOTAI]

Le 30/05/2012 14:44, Matthew J. Roth a écrit :
Considering that you made progress on your initial problem by setting 
nat=force_rport (resulting in connected calls with no audio) and now 
you're mentioning the use of externaddr, I'd recommend a very 
careful reading of the NAT SUPPORT section of sip.conf.sample in the 
configs directory of the Asterisk source tree.


I did read all those documentation, belive me. Also keep in mind that I 
*ONLY* face this problem with this provider, people using voipbuster or 
sipdiscount should have the same problem.


Concerning externaddr, this test server -dedicated to asterisk- being 
running in VM since ages, I never would suspect a NAT issue! Especially 
if previous 1.4 and 1.6 version are running smoothly ...


In Asterisk 1.8, there is a new configuration option named 
media_address which may be of particular interest.


media_address seems not an option, can be set only in general not per peer.

This is confusing because your first email said you had nat=no in 
your working 1.6.24 setup, but everything you're saying indicates a 
NAT problem to me.


Again, 1.6 version is perfectly working with this setup and conf files, 
and before 1.4 was too. And those both asterisk versions with *this* 
provider.


. A diagram showing all network elements between your Asterisk server 
and the remote host would be helpful.


Phone registration:

phone (Snom320 and GS GXV3175) - firewall1 (linux router) - Internet 
- firewall2 (linux router) - VM - phone account


Call:

phone account - Out of VM - firewall2 (linux router) - Internet - 
Peer IP - ???


To avoid further confusion, please include full and unaltered logs, 
SIP traces, and configurations in future posts.


During the time you and Andres replied to my post ;-) I got the same 
idea then him; and guess what, it's working! So problem is Asterisk 
1.8/10 in VM _only_ this provider(s) which are all Dellmont services.


Can someone confirm the problem?

Question is now, who is faulty? Should I open a bug?

Thanks for your time and support.
--
Daniel

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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Administrator TOOTAI]

Hi Matthew

Le 28/05/2012 19:28, Matthew J. Roth a écrit :

Administrator TOOTAI wrote:


we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
version and face the following problem: one of our peer
(voicetrading.com) doesn't accept our calls anymore, we receive a
timeout error Packet timed out after 32000ms with no response.

Switching back to 1.6 make things working again!

In sip.conf we have nat=no, peer conf is:


Asterisk 1.8.12 is not getting responses to the INVITES it sends.
Comparing the INVITES, the only significant difference I see is that
Asterisk 1.6.24 includes the rport field in the Via header and
Asterisk 1.8.12 does not:

   1.6.24 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
   1.8.12 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be

Try setting nat=force_rport in sip.conf.  Please reply back to the
list with the results.


We tested this setting this WE, effectively this problem disappear but 
another appears: call get connected but no audio. We installed Asterisk 
10.3.1 - connection and no audio too, so same behaviour.




There may be other differences between the versions that you haven't
accounted for.  Read the CHANGES and UPGRADE.txt files in the root of
the Asterisk source tree for details.


We did read those files, don't see which parameter we could have forget. 
media_address nor nat=comedia seems options for us. Hereunder a debug 
from call with force_rport: as you can see, the RTP audio is coming from 
another IP (77.77.777.77) We think asterisk doesn't accept this and 
don't know which parameter could solve this.



--- SIP read from UDP:111.111.1.111:5060 ---
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK04e390b0;rport
From: TOOTAi sip:00333@222.222.22.22;tag=as1335adb1
To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c31847cea
Contact: sip:0336@111.111.1.111:5060
Call-ID: 72d5d3df06c07cc6037786ee59f574df@222.222.22.22:5060
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 159

v=0
o=CARRIER 1338276550 1338276550 IN IP4 77.77.777.77
s=SIP Call
c=IN IP4 77.77.777.77
t=0 0
m=audio 41462 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
-
--- (11 headers 8 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw), peer - 
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), 
combined - 0x0 (nothing)

Peer audio RTP is at port 77.77.777.77:41462
list_route: hop: sip:0336@111.111.1.111:5060
set_destination: Parsing sip:0336@111.111.1.111:5060 for 
address/port to send to

set_destination: set destination to 111.111.1.111:5060
Transmitting (NAT) to 111.111.1.111:5060:
ACK sip:0336@111.111.1.111:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0d106caa;rport
Max-Forwards: 70
From: TOOTAi sip:00333@222.222.22.22;tag=as1335adb1
To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c31847cea
Contact: sip:00333@222.222.22.22:5060
Call-ID: 72d5d3df06c07cc6037786ee59f574df@222.222.22.22:5060
CSeq: 102 ACK
User-Agent: TOOTAiAudio
Content-Length: 0


---
-- SIP/myPeerDef-0003 answered SIP/104-0002

Thanks for your support.

--
Daniel

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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Matthew J. Roth]

Administrator TOOTAI wrote:

 We tested this setting this WE, effectively this problem disappear but 
 another appears: call get connected but no audio. We installed Asterisk 
 10.3.1 - connection and no audio too, so same behaviour.
 
 We did read those files, don't see which parameter we could have forget. 
 media_address nor nat=comedia seems options for us. Hereunder a debug 
 from call with force_rport: as you can see, the RTP audio is coming from 
 another IP (77.77.777.77) We think asterisk doesn't accept this and 
 don't know which parameter could solve this.


Daniel,

Asterisk is fine with RTP coming from another IP.  It used to work for
you on 1.6.24.  Here are the relevant bits from the 200 OK
responses:

  1.6.24 - o=CARRIER 1338135052 1338135052 IN IP4 77.72.168.74
   c=IN IP4 77.72.168.74
  1.8.12 - o=CARRIER 1338276550 1338276550 IN IP4 77.77.777.77
   c=IN IP4 77.77.777.77

Is that really the response that you received?  77.77.777.77 is not a
valid IP address (the 3rd octet is greater than 255), so if that's
what you're getting than your configuration is fine and the remote
end (or some proxy) is now the problem.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Administrator TOOTAI]

Le 29/05/2012 14:50, Matthew J. Roth a écrit :


Is that really the response that you received?  77.77.777.77 is not a
valid IP address (the 3rd octet is greater than 255), so if that's
what you're getting than your configuration is fine and the remote
end (or some proxy) is now the problem.



The IP address is valid, was 77.72.168.29 My bad with the caching stuff 
in the posted message, sorry.


I quit don't understand what happends: I reinstalled a fresh 1.6.24 
keeping the parameters from 1.8.13 and 10.3 version and it works! I 
again installed 10.3 and get again the


[2012-05-29 18:06:47] WARNING[17982]: chan_sip.c:3663 retrans_pkt: 
Retransmission timeout reached on transmission 
4fb581df7bc2f11f252f1ebe4718f264@10.0.70.12:5060 for seqno 102 (Critical 
Request) -- See 
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

10.0.70.12 is the IP of Asterisk server (kvm virtual machine) which is 
replaced by externaddr parameter from sip.conf.


I checked carefully a sip debug from the same call, same conf files, 
between 1.6.24 and 10.3.1: as with 1.6.24 I receive after the first 
INVITE a 183 Session progress, on the 10.3.1 I didn't receive it and 
Asterisk resend the INVITE. Despite this, the call is progressing, the 
phone on the other end is ringing but when answered, no audio, which 
seems normal.


My guess is that there is misunderstanding between Asterisk and the 
other end with 1.8.13/10.3 Will try with older version of 1.8 to see if 
problem is already there ...


If you have other ideas, welcome ;-)
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Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Matthew J. Roth]

Administrator TOOTAI wrote:

 we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
 version and face the following problem: one of our peer
 (voicetrading.com) doesn't accept our calls anymore, we receive a
 timeout error Packet timed out after 32000ms with no response.
 
 Switching back to 1.6 make things working again!
 
 In sip.conf we have nat=no, peer conf is:


Asterisk 1.8.12 is not getting responses to the INVITES it sends.
Comparing the INVITES, the only significant difference I see is that
Asterisk 1.6.24 includes the rport field in the Via header and
Asterisk 1.8.12 does not:

  1.6.24 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
  1.8.12 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be

Try setting nat=force_rport in sip.conf.  Please reply back to the
list with the results.

There may be other differences between the versions that you haven't
accounted for.  Read the CHANGES and UPGRADE.txt files in the root of
the Asterisk source tree for details.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

[Administrator TOOTAI]

Hi list,

we are upgrading our Asterisk production server from 1.6.24 to 1.8.12 
version and face the following problem: one of our peer 
(voicetrading.com) doesn't accept our calls anymore, we receive a 
timeout error Packet timed out after 32000ms with no response.


Switching back to 1.6 make things working again!

In sip.conf we have nat=no, peer conf is:

[myPeerDef]
type=peer
host=111.111.1.111
context=honeypot 



insecure=invite 



directmedia=no 



disallow=all 



allow=ulaw,alaw 



dtmfmode=inband

We aren't registered, our IP is authorized by their system.

Debug of sessions (222.222.22.22 is our server 111.111.1.111 is their)

Working one with 1.6:

Audio is at 222.222.22.22 port 26002
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 111.111.1.111:5060:
INVITE sip:0336@111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
Max-Forwards: 70
From: TOOTAi sip:00333@222.222.22.22;tag=as52190c5c
To: sip:0336@111.111.1.111
Contact: sip:00333@222.222.22.22
Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:10:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 284043376 284043376 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26002 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
-- Called myPeerDef/0336

--- SIP read from UDP:111.111.1.111:5060 ---
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
From: TOOTAi sip:00333@222.222.22.22;tag=as52190c5c
To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c3173f554
Contact: sip:0336@111.111.1.111:5060
Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 159

v=0
o=CARRIER 1338135040 1338135040 IN IP4 77.72.168.74
s=SIP Call
c=IN IP4 77.72.168.74
t=0 0
m=audio 18456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

-
--- (11 headers 8 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), 
combined - 0x0 (nothing)

Peer audio RTP is at port 77.72.168.74:18456
Peer doesn't provide video
-- SIP/myPeerDef-0007 is making progress passing it to 
SIP/104-0006


--- SIP read from UDP:111.111.1.111:5060 ---
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
From: TOOTAi sip:00333@222.222.22.22;tag=as52190c5c
To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c3173f554
Contact: sip:0336@111.111.1.111:5060
Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 159

v=0
o=CARRIER 1338135052 1338135052 IN IP4 77.72.168.74
s=SIP Call
c=IN IP4 77.72.168.74
t=0 0
m=audio 18456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

-
--- (11 headers 8 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), 
combined - 0x0 (nothing)

Peer audio RTP is at port 77.72.168.74:18456
Peer doesn't provide video
list_route: hop: sip:0336@111.111.1.111:5060
set_destination: Parsing sip:0336@111.111.1.111:5060 for 
address/port to send to

set_destination: set destination to 111.111.1.111, port 5060
Transmitting (no NAT) to 111.111.1.111:5060:
ACK sip:0336@111.111.1.111:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK5afa4cc0;rport
Max-Forwards: 70
From: TOOTAi sip:00333@222.222.22.22;tag=as52190c5c
To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c3173f554
Contact: sip:00333@222.222.22.22
Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22
CSeq: 102 ACK
User-Agent: TOOTAiAudio
Content-Length: 0


---
-- SIP/myPeerDef-0007 answered SIP/104-0006
Scheduling destruction of SIP dialog 
'2c974a0a2b08abe320ed388433e47d7e@222.222.22.22' in 32000 ms (Method: 
INVITE)
set_destination: Parsing sip:0336@111.111.1.111:5060 for 
address/port to send to

set_destination: set destination to 111.111.1.111, port 5060
Reliably Transmitting (no NAT) to 111.111.1.111:5060:
BYE 

Re: [asterisk-users] Upgrade and recompilation

[Barry L. Kline]

On 02/01/2011 12:34 PM, Harel Cohen wrote:

 As one with theoretical knowledge in programing, but never on Linux, I
 can understand terms and code structure but I don’t know:
 
 1. What shell commands (e.g. ./configure, make, make install etc.)
 should I run to recompile Asterisk (same version)?
 
 2. What shell commands should I run if I want to apply a change to
 source code?
 
 3. Is there a general guide on how to upgrade Asterisk?

Read the README file included with the source.

Barry

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Re: [asterisk-users] upgrade

[Elliot Murdock]

Hello!

You may want to check out http://linuxinnovations.com, a simple
reference describing the practical differences between the various
versions of Asterisk.  Seems it includes now version Asterisk 1.8.

--Elliot

On Sun, Nov 14, 2010 at 8:17 AM, Kyle Kienapfel doctor.w...@gmail.com wrote:


 On Sat, Nov 13, 2010 at 8:00 PM, Thomas Perron thomas.per...@gmail.com
 wrote:

 i am running 1.4.37 and am hosted on Rackspace.
 I feel like a took a step back by using the Cloud server service since
 I am having a little trouble proving that my basic configuration is
 working.
 Nevertheless, I want to upgrade to 1.8.
 I use Centos 5.5

 Anyone know of a good link that can help please?  I searched Google
 and got confused by the options.

 Upgrade to 1.8.  How please?

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 When asking questions here, you should try and provide some details that
 round out your setup.
 So its in a cloud, are you using just SIP or other stuff as well? Any AGI's?
 Why can't you prove that your basic configuration is working?
 Have you read the blurb about pipes vs commas in extensions.conf in regards
 to compatibilty with 1.2, 1.4 and 1.6? if not read the guide that explains
 differences between 1.4 and 1.6.
 Confused by what options?
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[asterisk-users] upgrade

[Thomas Perron]

i am running 1.4.37 and am hosted on Rackspace.
I feel like a took a step back by using the Cloud server service since
I am having a little trouble proving that my basic configuration is
working.
Nevertheless, I want to upgrade to 1.8.
I use Centos 5.5

Anyone know of a good link that can help please?  I searched Google
and got confused by the options.

Upgrade to 1.8.  How please?

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Re: [asterisk-users] upgrade

[Kyle Kienapfel]

On Sat, Nov 13, 2010 at 8:00 PM, Thomas Perron thomas.per...@gmail.comwrote:

 i am running 1.4.37 and am hosted on Rackspace.
 I feel like a took a step back by using the Cloud server service since
 I am having a little trouble proving that my basic configuration is
 working.
 Nevertheless, I want to upgrade to 1.8.
 I use Centos 5.5

 Anyone know of a good link that can help please?  I searched Google
 and got confused by the options.

 Upgrade to 1.8.  How please?

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When asking questions here, you should try and provide some details that
round out your setup.

So its in a cloud, are you using just SIP or other stuff as well? Any AGI's?

Why can't you prove that your basic configuration is working?

Have you read the blurb about pipes vs commas in extensions.conf in regards
to compatibilty with 1.2, 1.4 and 1.6? if not read the guide that explains
differences between 1.4 and 1.6.

Confused by what options?
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Re: [asterisk-users] upgrade 1.6 - 1.8: wrong password!

[pepesz]

Dear Paul,

I submitted the issue to the tracker.
ID 0018263

Thanks
pepesz

On Thu, Nov 4, 2010 at 8:46 PM, Paul Belanger
paul.belan...@polybeacon.comwrote:

 On Thu, Nov 4, 2010 at 3:24 PM, pepesz pep...@gmail.com wrote:
 snip
  WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
  nonce=5fcd5fa1   
 
 /snip
 I'm surprised to see the extra whitespaces in the nonce value.

  What can be the problem?
 
 If your working configuration worked with 1.6.2 but not 1.8, please
 created a new issue on the tracker and we will triage it.  Also
 include a debug log [1].

 [1]
 https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

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[asterisk-users] upgrade 1.6 - 1.8: wrong password!

[pepesz]

Dear All,

Today I upgraded asterisk 1.6 to 1.8.
As the result of this when peers trying to register to asterisk the system
shows:
NOTICE[24698]: chan_sip.c:23417 handle_request_register: Registration from
'50 sip:5...@192.168.1.109 sip:5...@192.168.1.109' failed for '
192.168.1.80:5062' - Wrong password
even though on 1.6 everything was OK

here is part of debug messages:
---cut---
--- Transmitting (NAT) to 192.168.1.50:5062 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.50:5062;branch=z9hG4bK-d8754z-d7545057f425cd49-1---d8754z-;received=192.168.1.50;rport=5062

From: 51sip:5...@192.168.1.109:5062 
sip:5...@192.168.1.109:5062;tag=172a701e

To: 51sip:5...@192.168.1.109:5062 
sip:5...@192.168.1.109:5062;tag=as6773fc96

Call-ID: ODlkNDYyMmYwNDAwYzIyMjEzOWZiYzMzNTRlNDhjNmQ.
CSeq: 3 REGISTER
Server: Asterisk PBX 1.8.0:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=5fcd5fa1   
Content-Length: 0
---cut---

and here the sip.conf part of that peer
---cut---
[50]
type=friend
defaultuser=50
secret=xx
context=4every1
callerid=Gigaset 50
host=dynamic
dtmfmode=rfc2833
---cut---



What can be the problem?
Can someone show me example of sip.conf with Digest authentication (send me
file or drop the link to website)?
Short info how to use digest?

Thanks a lot
Best regards,

pepesz
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Re: [asterisk-users] upgrade 1.6 - 1.8: wrong password!

[Paul Belanger]

On Thu, Nov 4, 2010 at 3:24 PM, pepesz pep...@gmail.com wrote:
snip
 WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
 nonce=5fcd5fa1   

/snip
I'm surprised to see the extra whitespaces in the nonce value.

 What can be the problem?

If your working configuration worked with 1.6.2 but not 1.8, please
created a new issue on the tracker and we will triage it.  Also
include a debug log [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

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Polybeacon | Consultant
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Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger

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Re: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtimemysql

[Andrew Thomas]

This is a problem with extconfig.conf - not your res_ or cdr_ ones.

In your case - extconfig.conf probably contained something like
'sippeers = mysql,MyDBase,sippeers'.  The 'problem' is that the middle
parameter is no longer for the database name - it is for the context in
res_mysql.conf.  So, the above now becomes 'sippeers =
mysql,general,sippeers'.  Give that a go...

Cheers
Andy


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 08 September 2010 15:48
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with
realtimemysql


Hello,

in asterisk 1.4.30 all realtime configurations go well.

In asterisk 1.6.2.11 the following appears on CLI :

[Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL RealTime: Invalid database specified: MyDBase (check
res_mysql.conf)
[Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL RealTime: Invalid database specified: MyDBase (check
res_mysql.conf)

res_mysql.conf :

[general]
dbhost = 127.0.0.1
dbname = MyDBase
dbuser = asterisk
dbpass = mysecret
dbport = 3306
dbsock = /tmp/mysql.sock
requirements=warn ; or createclose or createchar


What do I need to change to be conform asterisk 1.6 ?!

Reloading, restarting asterisk and restarting my CentOS-server all
doesn't help.


Jonas.


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[asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql

[Jonas Kellens]

Hello,

in asterisk 1.4.30 all realtime configurations go well.

In asterisk 1.6.2.11 the following appears on CLI :

[Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: 
MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
[Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: 
MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)


res_mysql.conf :

[general]
dbhost = 127.0.0.1
dbname = MyDBase
dbuser = asterisk
dbpass = mysecret
dbport = 3306
dbsock = /tmp/mysql.sock
requirements=warn ; or createclose or createchar


What do I need to change to be conform asterisk 1.6 ?!

Reloading, restarting asterisk and restarting my CentOS-server all 
doesn't help.



Jonas.
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Re: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql

[Gareth Blades]

Jonas Kellens wrote:
 Hello,
 
 in asterisk 1.4.30 all realtime configurations go well.
 
 In asterisk 1.6.2.11 the following appears on CLI :
 
 [Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: 
 MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
 [Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: 
 MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
 
 res_mysql.conf :
 
 [general]
 dbhost = 127.0.0.1
 dbname = MyDBase
 dbuser = asterisk
 dbpass = mysecret
 dbport = 3306
 dbsock = /tmp/mysql.sock
 requirements=warn ; or createclose or createchar
 
 
 What do I need to change to be conform asterisk 1.6 ?!
 
 Reloading, restarting asterisk and restarting my CentOS-server all 
 doesn't help.
 
 
 Jonas.
 

You have no entry called MyDBase in there.
Rename '[general]' to '[MyDBase]' and give it another go.

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Re: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql

[Jonas Kellens]

On 09/08/2010 04:50 PM, Gareth Blades wrote:
 Jonas Kellens wrote:

 Hello,

 in asterisk 1.4.30 all realtime configurations go well.

 In asterisk 1.6.2.11 the following appears on CLI :

 [Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
 MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
 [Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
 MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)

 res_mysql.conf :

 [general]
 dbhost = 127.0.0.1
 dbname = MyDBase
 dbuser = asterisk
 dbpass = mysecret
 dbport = 3306
 dbsock = /tmp/mysql.sock
 requirements=warn ; or createclose or createchar


 What do I need to change to be conform asterisk 1.6 ?!

 Reloading, restarting asterisk and restarting my CentOS-server all
 doesn't help.


 Jonas.

  
 You have no entry called MyDBase in there.
 Rename '[general]' to '[MyDBase]' and give it another go.


This did not work.

This is the message :

[Sep  8 17:08:43] ERROR[1843]: res_config_mysql.c:1575 mysql_reconnect: 
MySQL RealTime: Failed to connect database server MyDBase on 127.0.0.1 
(err 1045). Check debug for more info.
[Sep  8 17:08:43] ERROR[1843]: res_config_mysql.c:1575 mysql_reconnect: 
MySQL RealTime: Failed to connect database server MyDBase on 127.0.0.1 
(err 1045). Check debug for more info.


However my log /var/log/asterisk/debug has no entry of today... Which 
debug info can I check then ?!


Jonas.




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Re: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql

[Gareth Blades]

Jonas Kellens wrote:
 On 09/08/2010 04:50 PM, Gareth Blades wrote:
 Jonas Kellens wrote:

 Hello,

 in asterisk 1.4.30 all realtime configurations go well.

 In asterisk 1.6.2.11 the following appears on CLI :

 [Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
 MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
 [Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
 MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)

 res_mysql.conf :

 [general]
 dbhost = 127.0.0.1
 dbname = MyDBase
 dbuser = asterisk
 dbpass = mysecret
 dbport = 3306
 dbsock = /tmp/mysql.sock
 requirements=warn ; or createclose or createchar


 What do I need to change to be conform asterisk 1.6 ?!

 Reloading, restarting asterisk and restarting my CentOS-server all
 doesn't help.


 Jonas.

  
 You have no entry called MyDBase in there.
 Rename '[general]' to '[MyDBase]' and give it another go.

 
 This did not work.
 
 This is the message :
 
 [Sep  8 17:08:43] ERROR[1843]: res_config_mysql.c:1575 mysql_reconnect: 
 MySQL RealTime: Failed to connect database server MyDBase on 127.0.0.1 
 (err 1045). Check debug for more info.
 [Sep  8 17:08:43] ERROR[1843]: res_config_mysql.c:1575 mysql_reconnect: 
 MySQL RealTime: Failed to connect database server MyDBase on 127.0.0.1 
 (err 1045). Check debug for more info.
 
 
 However my log /var/log/asterisk/debug has no entry of today... Which 
 debug info can I check then ?!
 
 
 Jonas.
 
 
 
 
You are closer.
1) Make sure the myDBase database exists and the asterisk user has 
permission to access it.
2) You have an IP address  port together with a socket filename listed. 
Decide whhich method you want to use and remove the configuration for 
the other.

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Re: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql

[Jonas Kellens]

On 09/08/2010 05:18 PM, Gareth Blades wrote:
 Jonas Kellens wrote:

 On 09/08/2010 04:50 PM, Gareth Blades wrote:
  
 Jonas Kellens wrote:


 Hello,

 in asterisk 1.4.30 all realtime configurations go well.

 In asterisk 1.6.2.11 the following appears on CLI :

 [Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
 MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
 [Sep  8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
 MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)

 res_mysql.conf :

 [general]
 dbhost = 127.0.0.1
 dbname = MyDBase
 dbuser = asterisk
 dbpass = mysecret
 dbport = 3306
 dbsock = /tmp/mysql.sock
 requirements=warn ; or createclose or createchar


 What do I need to change to be conform asterisk 1.6 ?!

 Reloading, restarting asterisk and restarting my CentOS-server all
 doesn't help.


 Jonas.


  
 You have no entry called MyDBase in there.
 Rename '[general]' to '[MyDBase]' and give it another go.


 This did not work.

 This is the message :

 [Sep  8 17:08:43] ERROR[1843]: res_config_mysql.c:1575 mysql_reconnect:
 MySQL RealTime: Failed to connect database server MyDBase on 127.0.0.1
 (err 1045). Check debug for more info.
 [Sep  8 17:08:43] ERROR[1843]: res_config_mysql.c:1575 mysql_reconnect:
 MySQL RealTime: Failed to connect database server MyDBase on 127.0.0.1
 (err 1045). Check debug for more info.


 However my log /var/log/asterisk/debug has no entry of today... Which
 debug info can I check then ?!


 Jonas.




  
 You are closer.
 1) Make sure the myDBase database exists and the asterisk user has
 permission to access it.
 2) You have an IP address  port together with a socket filename listed.
 Decide whhich method you want to use and remove the configuration for
 the other.



Created another user on the MySQL-DB and now it works...

Thanks.

Jonas.

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[asterisk-users] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call

[Giorgio Incantalupo]

Hi,

just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same 
zaptel/libpri/mISDN/add-ons.
It crashes when transferring a call.
Anybody tried it with success?

Thank you

Giorgio

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Re: [asterisk-users] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call

[Jonn Taylor]

Giorgio Incantalupo wrote:
 Hi,

 just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same 
 zaptel/libpri/mISDN/add-ons.
 It crashes when transferring a call.
 Anybody tried it with success?

 Thank you

 Giorgio

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You really should upgrade all of them. But you have to do add-ons!

-- 
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Taylor Telephone Systems, Inc
8334 Argenta Trail
Inver Grove Heights, MN 55077

http://www.taylortelephone.com/



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Re: [asterisk-users] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call

[Mark Michelson]

Giorgio Incantalupo wrote:
 Hi,
 
 just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same 
 zaptel/libpri/mISDN/add-ons.
 It crashes when transferring a call.
 Anybody tried it with success?
 
 Thank you
 
 Giorgio
 

If you're having crashes occur when transferring a call, you should report it 
as 
a bug on bugs.digium.com. Be sure to attach a backtrace from the crash as 
described in doc/backtrace.txt in the Asterisk source.

Thanks,
Mark Michelson

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Re: [asterisk-users] Upgrade Cisco 7971G-GE from SCCP to SIP

[César García]

Ayman, after you BUY the license/firmware, etc,  to cisco, I use 7911G with
Astterisk, my xml conf file is in the wiki

: )

2009/1/13 Steve Edwards asterisk@sedwards.com

 On Tue, 13 Jan 2009, Ayman Boules (Live.COM) wrote:

  It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP.
  If so, please email me the detailed instructions to do the upgrade.

 Where's that link to http://letmegogglethatforyou.com?;

  I will appreciate it much if you have the latest 8.4(2) firmware (file
  name: cmterm-7970_7971-sip.8-4-2.cop) and email it to me or send me a
  link to download it...

 Oh. Of course. Let's all violate cisco's copyright on a public mailing
 list :)

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

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Re: [asterisk-users] Upgrade to v.1.2.31 ... weird change

[Leonardo Gomes Figueira]

Tilghman Lesher escreveu:
 On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
 I think it happened when I upgraded an install to 1.2.31

 The variable CALLERIDNUM no longer works and CallerID(num) has to be
 used.
 
 I don't see why not.  There has been no change whatsoever to that body of
 code.

I think there is some mistake on his test about CALLERIDNUM. Did a quick
test here on 1.2.31 and it's working fine.

On the other hand, the change in chan_iax2.c on 1.2.31 really broke
something... but it's not related to dialplan variables.

IAX2 peer registration:

http://bugs.digium.com/view.php?id=14238

  Leonardo


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Re: [asterisk-users] Upgrade to v.1.2.31 ... weird change

[Steve Kennedy]

On Wed, Jan 14, 2009 at 02:56:44PM -0200, Leonardo Gomes Figueira wrote:

 Tilghman Lesher escreveu:
  On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
  I think it happened when I upgraded an install to 1.2.31
  The variable CALLERIDNUM no longer works and CallerID(num) has to be
  used.
  I don't see why not.  There has been no change whatsoever to that body of
  code.
 I think there is some mistake on his test about CALLERIDNUM. Did a quick
 test here on 1.2.31 and it's working fine.

I'll check, but something definately changed. I think I was using
${CALLERIDNUM:1:4} anyway didn't work as planned. Changing to
CallerID(num) and it worked again.


Steve

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Re: [asterisk-users] Upgrade Cisco 7971G-GE from SCCP to SIP

[Ayman Boules (Live.COM)]

Good Morning Everyone,

It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP.  If
so, please email me the detailed instructions to do the upgrade.

I will appreciate it much if you have the latest 8.4(2) firmware (file name:
cmterm-7970_7971-sip.8-4-2.cop) and email it to me or send me a link to
download it...



Regards,

Ayman L. Boules
Sunday, January 11, 2009
++

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason
(Lists)
Sent: Sunday, January 11, 2009 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_sip on non-standard port 5062 - contact
has no port

You are configuring Asterisk to LISTEN on 5062 , if you want it to talk 
to another server on 5062, then configure that server's config stanza 
accordingly.

-- 
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Re: [asterisk-users] Upgrade Cisco 7971G-GE from SCCP to SIP

[Steve Edwards]

On Tue, 13 Jan 2009, Ayman Boules (Live.COM) wrote:

 It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP. 
 If so, please email me the detailed instructions to do the upgrade.

Where's that link to http://letmegogglethatforyou.com?;

 I will appreciate it much if you have the latest 8.4(2) firmware (file 
 name: cmterm-7970_7971-sip.8-4-2.cop) and email it to me or send me a 
 link to download it...

Oh. Of course. Let's all violate cisco's copyright on a public mailing 
list :)

Thanks in advance,

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

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[asterisk-users] Upgrade to v.1.2.31 ... weird change

[Steve Kennedy]

I think it happened when I upgraded an install to 1.2.31

The variable CALLERIDNUM no longer works and CallerID(num) has to be
used.

I know the initial one was being depreciated, but I didn't see any
mention of it.

Steve

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Re: [asterisk-users] Upgrade to v.1.2.31 ... weird change

[Tilghman Lesher]

On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
 I think it happened when I upgraded an install to 1.2.31

 The variable CALLERIDNUM no longer works and CallerID(num) has to be
 used.

I don't see why not.  There has been no change whatsoever to that body of
code.

 I know the initial one was being depreciated, but I didn't see any
 mention of it.

I think you mean deprecated.  Depreciation is an accounting term.

-- 
Tilghman

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Re: [asterisk-users] Upgrade to v.1.2.31 ... weird change

[Anthony Francis]

Tilghman Lesher wrote:
 On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
   
 I think it happened when I upgraded an install to 1.2.31

 The variable CALLERIDNUM no longer works and CallerID(num) has to be
 used.
 

 I don't see why not.  There has been no change whatsoever to that body of
 code.

   
 I know the initial one was being depreciated, but I didn't see any
 mention of it.
 

 I think you mean deprecated.  Depreciation is an accounting term.

   
The old variables for callerid where indeed put on the chapping block of 
deprecation, if you turn your cli verbosity to 3 or higher you should 
see warnings everytime the old variable is used.

Anthony

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Re: [asterisk-users] Upgrade Cisco 7971G-GE from SCCP to SIP

[Ayman Boules (Live.COM)]

Good Morning Everyone,

It will be great if someone can help me upgrade a Cisco 7971G-GE to SIP.  If
so, please email me the detailed instructions to do the upgrade.

I will appreciate it much if you have the latest 8.4(2) firmware (file name:
cmterm-7970_7971-sip.8-4-2.cop) and email it to me or send me a link to
download it...



Regards,

Ayman L. Boules
Sunday, January 11, 2009
++

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason
(Lists)
Sent: Sunday, January 11, 2009 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_sip on non-standard port 5062 - contact
has no port

You are configuring Asterisk to LISTEN on 5062 , if you want it to talk 
to another server on 5062, then configure that server's config stanza 
accordingly.

-- 
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[asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

[Jerry Geis]

Hi all,

I upgraded from asterisk 1.2.23 and zaptel 1.2.19
to asterisk 1.4.18 and zaptel 1.4.12.1
I use polycom 501 phones internally.

Everything seems fine. I can pick up the phone and call out,
calls coming in work just fine.

The issue I see is when the system first calls me,
then calls someone else. This works if its polycom to polycom. I hear 
audio full channel.
If I do  polycom to external line like a cell I only get HALF channel audio.
At this time they can hear me but I cannot hear them.

What might this be???

Jerry

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Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

[Danny Nicholas]

You could trying changing this in sip.cfg
AES voice.aes.hs.enable=0 
To
AES voice.aes.hs.enable=1 

It's at line 324 in mine.  Results not guaranteed.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 10:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

Hi all,

I upgraded from asterisk 1.2.23 and zaptel 1.2.19
to asterisk 1.4.18 and zaptel 1.4.12.1
I use polycom 501 phones internally.

Everything seems fine. I can pick up the phone and call out,
calls coming in work just fine.

The issue I see is when the system first calls me,
then calls someone else. This works if its polycom to polycom. I hear 
audio full channel.
If I do  polycom to external line like a cell I only get HALF channel audio.
At this time they can hear me but I cannot hear them.

What might this be???

Jerry

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Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

[Jerry Geis]

 You could trying changing this in sip.cfg
 AES voice.aes.hs.enable=0 
 To
 AES voice.aes.hs.enable=1 

   

Just tried that - rebooted my polycom and still half audio.
Thanks,

Jerry

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Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half channelaudio

[Danny Nicholas]

You could try un-commenting duplex=2 in rpt.conf and changing it to
duplex=3.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 11:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half
channelaudio


 You could trying changing this in sip.cfg
 AES voice.aes.hs.enable=0 
 To
 AES voice.aes.hs.enable=1 

   

Just tried that - rebooted my polycom and still half audio.
Thanks,

Jerry

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Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

[Jerry Geis]

I am using an AGI to setup the call to the first person,
then jumping into the dialplan with some Variables set.

Is the AGI messing up my channel???

My dialplan at that point looks like:

exten = 
call_cont,1,Dial(${CONT_CALLAT},${CONT_DIAL_TIMEOUT},${CONT_ONHOLD}tT)

CONT_CALLAT is Zap/1/506 where X is my number
CONT_DIAL_TIMEOUT is 60
CONT_ONHOLD is tT

Seems like this should still be working also.
How do I tell where/how my audio is getting blocked. Internal polycom to 
polycom works fine with this AGI,
the old 1.2 worked fine with this AGI, its just polycom to external 
world with the AGI is giving me a half channel.

Jerry


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[asterisk-users] Upgrade 1.4.19 to 1.6 = segementation fault

[Ronald Wiplinger (Lists)]

During compiling I have not seen an error, however, when I start
asterisk again it ends with:


app_morsecode.so = (Morse code)
  == Registered custom function 'SYSINFO'
 func_sysinfo.so = (System information related functions)
Segmentation fault (core dumped)


How can I figure out what is wrong?

bye

Ronald

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Re: [asterisk-users] upgrade to 1.6

[Hakan C]

Can you post your dialplan?

On Mon, Nov 17, 2008 at 4:41 AM, Jerry Geis [EMAIL PROTECTED] wrote:

 When I upgraded from 1.4 to 1.6 it seems my dialplan extensions.conf
 is not loaded. in UPGRADE.txt I dont see any reason why.

 ^[[1;30m  == ^[[0mParsing '/etc/asterisk/extensions.conf': ^[[1;30m  ==
 ^[[0mFound

 It shows its parsing with no errors.

 dialplan show - does not show anything from extentions.conf
 It only shows ael stuff.

 I am confused?

 jerry

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Re: [asterisk-users] upgrade to 1.6

[Jerry Geis]

Jerry Geis wrote:
 When I upgraded from 1.4 to 1.6 it seems my dialplan extensions.conf
 is not loaded. in UPGRADE.txt I dont see any reason why.

 ^[[1;30m  == ^[[0mParsing '/etc/asterisk/extensions.conf': ^[[1;30m  
 == ^[[0mFound

 It shows its parsing with no errors.

 dialplan show - does not show anything from extentions.conf
 It only shows ael stuff.

 I am confused?

 jerry

I think I have narrowed it down to doing a:
#include
for a file that is not present.

There was no error message on the console about it that I saw.
However I removed more and more lines until something loaded.
after that I narrowed it down to that #includes.

Jerry


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Re: [asterisk-users] upgrade to 1.6

[Matt Riddell]

On 18/11/2008 2:44 a.m., Jerry Geis wrote:
 Jerry Geis wrote:
 When I upgraded from 1.4 to 1.6 it seems my dialplan extensions.conf
 is not loaded. in UPGRADE.txt I dont see any reason why.

 ^[[1;30m  == ^[[0mParsing '/etc/asterisk/extensions.conf': ^[[1;30m  
 == ^[[0mFound

 It shows its parsing with no errors.

 dialplan show - does not show anything from extentions.conf
 It only shows ael stuff.

 I am confused?

 jerry

 I think I have narrowed it down to doing a:
 #include
 for a file that is not present.
 
 There was no error message on the console about it that I saw.
 However I removed more and more lines until something loaded.
 after that I narrowed it down to that #includes.

In the latest versions of Asterisk, including a file that does not exist
causes the entire config file to not be read.

-- 
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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[asterisk-users] upgrade to 1.6

[Jerry Geis]

When I upgraded from 1.4 to 1.6 it seems my dialplan extensions.conf
is not loaded. in UPGRADE.txt I dont see any reason why.

^[[1;30m  == ^[[0mParsing '/etc/asterisk/extensions.conf': ^[[1;30m  == 
^[[0mFound

It shows its parsing with no errors.

dialplan show - does not show anything from extentions.conf
It only shows ael stuff.

I am confused?

jerry

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[asterisk-users] upgrade of asterisk .... to what?

[nik600]

Hi to all

i'm managing a call center with 20 operators using Asterisk.

I'm still using Asterisk 1.2.x as i love his stability.

Now, i'm planning to migrate to 1.4.x, but i don't know what version
to use! 1.4.20 has been released a few days ago, but now there is
1.4.21.

Is there a rule to determine what is beta and what is stable?

Thanks

-- 
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser

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Re: [asterisk-users] upgrade of asterisk .... to what?

[Alex Balashov]

nik600 wrote:

 Hi to all
 
 i'm managing a call center with 20 operators using Asterisk.
 
 I'm still using Asterisk 1.2.x as i love his stability.
 
 Now, i'm planning to migrate to 1.4.x, but i don't know what version
 to use! 1.4.20 has been released a few days ago, but now there is
 1.4.21.
 
 Is there a rule to determine what is beta and what is stable?

If something is officially released, it is considered stable, from a 
prescriptive point of view.

Whether that proves to be the case in fact... is a matter of 
considerable controversy and objective variation.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] upgrade of asterisk .... to what?

[Gordon Henderson]

On Thu, 22 May 2008, nik600 wrote:

 Hi to all

 i'm managing a call center with 20 operators using Asterisk.

 I'm still using Asterisk 1.2.x as i love his stability.

 Now, i'm planning to migrate to 1.4.x, but i don't know what version
 to use! 1.4.20 has been released a few days ago, but now there is
 1.4.21.

Are there any features in 1.4 that you desperately need? If not, then why 
upgrade?

Gordon
(still running 1.2.x)

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Re: [asterisk-users] upgrade of asterisk .... to what?

[nik600]

On Thu, May 22, 2008 at 12:31 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
 On Thu, 22 May 2008, nik600 wrote:

 Hi to all

 i'm managing a call center with 20 operators using Asterisk.

 I'm still using Asterisk 1.2.x as i love his stability.

 Now, i'm planning to migrate to 1.4.x, but i don't know what version
 to use! 1.4.20 has been released a few days ago, but now there is
 1.4.21.

 Are there any features in 1.4 that you desperately need? If not, then why
 upgrade?

No, i'm just wondering because there is creating a greater difference
between my installation and the actual Asterisk.

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Re: [asterisk-users] upgrade of asterisk .... to what?

[Brian J. Murrell]

On Thu, 2008-05-22 at 14:05 +0200, nik600 wrote:
 
 No, i'm just wondering because there is creating a greater difference
 between my installation and the actual Asterisk.

If it ain't broke, don't fix it.  You are already so far behind that any
upgrade is going to be a major task of testing and verification on your
part, so why not just wait until you have an actual reason to upgrade.

As long as it's being supported and does it's job for you, I'd just stay
right where you are.

b.



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Re: [asterisk-users] upgrade of asterisk .... to what?

[Sherwood McGowan]

Brian J. Murrell wrote:
 On Thu, 2008-05-22 at 14:05 +0200, nik600 wrote:
   
 No, i'm just wondering because there is creating a greater difference
 between my installation and the actual Asterisk.
 

 If it ain't broke, don't fix it.  You are already so far behind that any
 upgrade is going to be a major task of testing and verification on your
 part, so why not just wait until you have an actual reason to upgrade.

 As long as it's being supported and does it's job for you, I'd just stay
 right where you are.

 b.

   
 

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He's right, I have a client who is running 1.2 but wishes to upgrade and 
it's going to be a pretty large undertaking. If nothing else, look at 
the change logs for 1.4 AND 1.6 and then decide if you're in need of an 
upgrade.

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Re: [asterisk-users] upgrade of asterisk .... to what?

[Benny Amorsen]

Sherwood McGowan [EMAIL PROTECTED] writes:

 He's right, I have a client who is running 1.2 but wishes to upgrade and 
 it's going to be a pretty large undertaking. If nothing else, look at 
 the change logs for 1.4 AND 1.6 and then decide if you're in need of an 
 upgrade.

In practice, upgrades to 1.4 are not all that difficult unless you
areusing many deprecated things in 1.2. All we tend to hit is that
queue members defined in queues.conf are permanent --
RemoveQueueMember does not remove them.

I upgraded my home Asterisk to 1.6 beta, and the only thing broken so
far was a Dial command with | used as a separator. I was rather
impressed.


/Benny



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Re: [asterisk-users] Upgrade 1.2 - 1.4 voice files

[Phil Knighton]

Hi Adrian

I'm using 1.4.10, and all of my voicemail sound files are in SLN, so you
should be able to use them without a problem.  Just check the notes for
the slightly different location for sound files under 1.4.

Thanks

Phil

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian
Marsh
Sent: 08 February 2008 14:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Upgrade 1.2 - 1.4 voice files

Hi All,

I'm going to be upgrading our 1.2 Asterisk system. At the moment we use
the Enicomms SLN files.  Are there major differences in the 1.4 default
voicefile packs, or will I be able to re-use Enicomms??

In the Make menuselect, I noticed theres no .SLN voicefile selection for
the basic audiofiles - has SLN been depreciated?

Thanks

Adrian

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Re: [asterisk-users] Upgrade 1.2 - 1.4 voice files

[Russell Bryant]

Adrian Marsh wrote:
 In the Make menuselect, I noticed theres no .SLN voicefile selection for
 the basic audiofiles - has SLN been depreciated?

No, the sln format is still supported.  We have just never distributed any 
files 
in that raw format.  Previously, we only had gsm recordings.  For Asterisk 1.4, 
we got all of the prompts re-recorded so that we could distribute them in a 
number of higher-quality codecs, as well as in 3 languages.

The actually scripts of the files has not changed much, as far as I remember. 
The sounds.txt file in 1.2, and the 1.4 sounds packages should say exactly what 
they are.  You can always compare them with diff.

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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[asterisk-users] Upgrade 1.2 - 1.4 voice files

[Adrian Marsh]

Hi All,

I'm going to be upgrading our 1.2 Asterisk system. At the moment we use
the Enicomms SLN files.  Are there major differences in the 1.4 default
voicefile packs, or will I be able to re-use Enicomms??

In the Make menuselect, I noticed theres no .SLN voicefile selection for
the basic audiofiles - has SLN been depreciated?

Thanks

Adrian

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[asterisk-users] Upgrade fails, need system upgrade advice

[Ronald Wiplinger]

I have a AMD64 CPU and use SuSE 9.2 with kernel 2.6.8-18
I tried to upgrade svn version 1.4.x but it fails at each part and
mainly because the system is with 1100 days getting to old.

I have to make a decision and need your advice.

CPU AMD64 3200+
1 GB RAM
Digium card with 2 FXS and 2 FXO
external Wellgate box 3804

I want to keep my current settings (backup /etc/asterisk and
/var/lib/asterisk and /var/spool/asterisk)
I use festiva
I need multiple fax on different extensions
I would like to run also OpenSer on the same machine


I would like to re-install a new system with svn asterisk 1.4.x and the
above settings.
Would you suggest me to install
a. OpenSuse 10.x
b. Ubuntu desktop
c. Ubuntu server

Any other hints? to backup directories? or just use a new hard disk.
With LVM?

bye

Ronald

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Philipp Kempgen]

Tilghman Lesher wrote:
 On Monday 24 December 2007 10:30:57 Dovid B wrote:

 While this encourages me to use 1.4 at the same time it makes me wonder why
 Digium waited that long...
 
 Because IT has other things to do than upgrade the PBX?

Which makes for a good answer to Olle's original question. :)


Merry Christmas,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? - http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Anthony Francis]

Axel Thimm wrote:
 On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote:
   
 Olle E Johansson [EMAIL PROTECTED] writes:

 
 But on the other hand, if people rely on third-party distributions
 we might want to set up some kind of peer pressure on the
 maintainers - and possibly identify them so we can support them and
 speed up their process.
   
 Third-party distributions are very important, and Asterisk has
 for various reasons done relatively badly there.

 Fedora still doesn't have Asterisk, but does have CallWeaver. Asterisk
 isn't even available in the most popular extra repositories, but only
 in ATrpms, my least favourite of the larger repositories.
 

 It happens to be my favourite thrid party repo though, ;) and indeed
 there is quite some asterisk support happening there.
   
 

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Asterisk is fairly easy to build, I don't see why it needs to be in a 
repo. IMO

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[MatsK]

Anthony Francis wrote:
 Axel Thimm wrote:
 On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote:
   
 Olle E Johansson [EMAIL PROTECTED] writes:

 
 But on the other hand, if people rely on third-party distributions
 we might want to set up some kind of peer pressure on the
 maintainers - and possibly identify them so we can support them and
 speed up their process.
   
 Third-party distributions are very important, and Asterisk has
 for various reasons done relatively badly there.

 Fedora still doesn't have Asterisk, but does have CallWeaver. Asterisk
 isn't even available in the most popular extra repositories, but only
 in ATrpms, my least favourite of the larger repositories.
 
 It happens to be my favourite thrid party repo though, ;) and indeed
 there is quite some asterisk support happening there.
   
 

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 Asterisk is fairly easy to build, I don't see why it needs to be in a 
 repo. IMO

There are several benefits to have it in a repo.
One is that it is a security issue, you don't want to have dev tools on
a exposed server.
Another is, if you have hundreds of similar machines, why compile
Asterisk 100 times when you need to compile it once and then just copy
the binaries to the other 99 machines.

So as you see it is an advantage with repo's.


Merry Christmas
Mats


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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Tzafrir Cohen]

On Mon, Dec 24, 2007 at 04:11:30AM -0700, Anthony Francis wrote:

 Asterisk is fairly easy to build, I don't see why it needs to be in a 
 repo. IMO

Why does it need to be in a tarball? Isn't it simpler to just grab from
an SVN tag?

There are many benefits to a reproducable build. Also consider that
Asterisk is often part if a bigger product. Asterisk is essentially not
a PBX, but rather a PBX building toolkit. It is very customizable and
can do many things. And therefore can be integrated in many products.

One of those products is a binary distribution by Digium: AsteriskNow. 

So it seems that some others do see the need. I suggest not to start 
YAHW on that subject :-)

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

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Patrick]

On Mon, 2007-12-24 at 04:11 -0700, Anthony Francis wrote:
 Axel Thimm wrote:
  On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote:

  Olle E Johansson [EMAIL PROTECTED] writes:
 
  
  But on the other hand, if people rely on third-party distributions
  we might want to set up some kind of peer pressure on the
  maintainers - and possibly identify them so we can support them and
  speed up their process.

  Third-party distributions are very important, and Asterisk has
  for various reasons done relatively badly there.
 
  Fedora still doesn't have Asterisk, but does have CallWeaver. Asterisk
  isn't even available in the most popular extra repositories, but only
  in ATrpms, my least favourite of the larger repositories.
  
 
  It happens to be my favourite thrid party repo though, ;) and indeed
  there is quite some asterisk support happening there.

[snip]

 Asterisk is fairly easy to build, I don't see why it needs to be in a 
 repo. IMO

For example because you don't have a build environment (gcc, autoconf
etc.) on a production box. A repo allows you to build on one box and
deploy the RPMs via the repo on the other boxes.

Regards,
Patrick


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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Hans Witvliet]

On Mon, 2007-12-24 at 04:11 -0700, Anthony Francis wrote:
 Axel Thimm wrote:
  On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote:

  Olle E Johansson [EMAIL PROTECTED] writes:
 
  
  But on the other hand, if people rely on third-party distributions
  we might want to set up some kind of peer pressure on the
  maintainers - and possibly identify them so we can support them and
  speed up their process.

  Third-party distributions are very important, and Asterisk has
  for various reasons done relatively badly there.
 
  Fedora still doesn't have Asterisk, but does have CallWeaver. Asterisk
  isn't even available in the most popular extra repositories, but only
  in ATrpms, my least favourite of the larger repositories.
  
 
  It happens to be my favourite thrid party repo though, ;) and indeed
  there is quite some asterisk support happening there.


 Asterisk is fairly easy to build, I don't see why it needs to be in a 
 repo. IMO
 
 ___

Such as:
http://ftp5.gwdg.de/pub/opensuse/repositories/network:/telephony/openSUSE_10.3/
...


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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Dovid B]

- Original Message - 
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Friday, December 21, 2007 9:56 PM
Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!


 On Friday 21 December 2007 13:16:17 Matt wrote:
 It may be a year old.. but until Digium is drinking their own dog food.. 
 I
 won't be using it.

 I beg your pardon.  The Digium IVR has been on 1.4 since about April or 
 so.

While this encourages me to use 1.4 at the same time it makes me wonder why 
Digium waited that long... 



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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Tilghman Lesher]

On Monday 24 December 2007 10:30:57 Dovid B wrote:
 Tilghman Lesher wrote:
  On Friday 21 December 2007 13:16:17 Matt wrote:
  It may be a year old.. but until Digium is drinking their own dog food..
  I
  won't be using it.
 
  I beg your pardon.  The Digium IVR has been on 1.4 since about April or
  so.

 While this encourages me to use 1.4 at the same time it makes me wonder why
 Digium waited that long...

Because IT has other things to do than upgrade the PBX?

-- 
Tilghman

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Axel Thimm]

On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote:
 Olle E Johansson [EMAIL PROTECTED] writes:
 
  But on the other hand, if people rely on third-party distributions
  we might want to set up some kind of peer pressure on the
  maintainers - and possibly identify them so we can support them and
  speed up their process.
 
 Third-party distributions are very important, and Asterisk has
 for various reasons done relatively badly there.
 
 Fedora still doesn't have Asterisk, but does have CallWeaver. Asterisk
 isn't even available in the most popular extra repositories, but only
 in ATrpms, my least favourite of the larger repositories.

It happens to be my favourite thrid party repo though, ;) and indeed
there is quite some asterisk support happening there.
-- 
Axel.Thimm at ATrpms.net


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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Remco Barendse]

 I wonder if there are any major obstacles for upgrading.


Just tried an in-place upgrade on my home box :

make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so 
res_config_mysql.so; do /usr/bin/install -c -m 755 $x 
/usr/lib/asterisk/modules ; done
/usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or 
directory
/usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or 
directory
/usr/bin/install: cannot stat `res_config_mysql.so': No such file or 
directory
make: *** [install] Error 1


And the asterisk console is flooded with these errors :

[Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 
determine_firstline_parts: Bad request protocol Packet
[Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 
determine_firstline_parts: Bad request protocol Packet

So for the next time to come i'll turn back to 1.2 :)

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[dave cantera]

remco,
I just had the same problem/error on my CLI when I added a polycom
shoretel IP-100 phone to my network and enabled mgcp... couldn't
figure out how to get that working yet... 
I don't think it is related to 1.4 as I have been running 1.4 has been
running for over a year now without that error... I would look
somewhere else...
daveC


[Dec 21 08:51:32] WARNING[16742]: chan_sip.c:6620
determine_firstline_parts: Bad request protocol
[EMAIL PROTECTED]] MGCP 1.0


Remco Barendse wrote:

  
I wonder if there are any major obstacles for upgrading.

  
  

Just tried an in-place upgrade on my home box :

make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so 
res_config_mysql.so; do /usr/bin/install -c -m 755 $x 
/usr/lib/asterisk/modules ; done
/usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or 
directory
/usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or 
directory
/usr/bin/install: cannot stat `res_config_mysql.so': No such file or 
directory
make: *** [install] Error 1


And the asterisk console is flooded with these errors :

[Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 
determine_firstline_parts: Bad request protocol Packet
[Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 
determine_firstline_parts: Bad request protocol Packet

So for the next time to come i'll turn back to 1.2 :)

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-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894






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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Matt]

It may be a year old.. but until Digium is drinking their own dog food.. I
won't be using it.

On Dec 21, 2007 9:26 AM, dave cantera [EMAIL PROTECTED] wrote:

  remco,
 I just had the same problem/error on my CLI  when I added a polycom
 shoretel IP-100 phone to my network and enabled mgcp...  couldn't figure out
 how to get that working yet...
 I don't think it is related to 1.4 as I have been running 1.4 has been
 running for over a year now without that error...  I would look somewhere
 else...
 daveC


 [Dec 21 08:51:32] WARNING[16742]: chan_sip.c:6620
 determine_firstline_parts: Bad request protocol
 [EMAIL PROTECTED] [EMAIL PROTECTED]]
 MGCP 1.0



 Remco Barendse wrote:

  I wonder if there are any major obstacles for upgrading.


  Just tried an in-place upgrade on my home box :

 make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
 for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so
 res_config_mysql.so; do /usr/bin/install -c -m 755 $x
 /usr/lib/asterisk/modules ; done
 /usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or
 directory
 /usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or
 directory
 /usr/bin/install: cannot stat `res_config_mysql.so': No such file or
 directory
 make: *** [install] Error 1


 And the asterisk console is flooded with these errors :

 [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707
 determine_firstline_parts: Bad request protocol Packet
 [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707
 determine_firstline_parts: Bad request protocol Packet

 So for the next time to come i'll turn back to 1.2 :)

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 --
 My wife's sister is in California.
 I should buy her a Videophone2008!

 Truly, The Next Best Thing to Being There!
 --
 WorldWideVideoPhones.com856.380.0894


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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Tilghman Lesher]

On Friday 21 December 2007 13:16:17 Matt wrote:
 It may be a year old.. but until Digium is drinking their own dog food.. I
 won't be using it.

I beg your pardon.  The Digium IVR has been on 1.4 since about April or so.

-- 
Tilghman

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Johansson Olle E]

21 dec 2007 kl. 10.12 skrev Remco Barendse:


 I wonder if there are any major obstacles for upgrading.


 Just tried an in-place upgrade on my home box :

 make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
 for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so
 res_config_mysql.so; do /usr/bin/install -c -m 755 $x
 /usr/lib/asterisk/modules ; done
 /usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file  
 or
 directory
 /usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or
 directory
 /usr/bin/install: cannot stat `res_config_mysql.so': No such file or
 directory
 make: *** [install] Error 1
For some reason, the mysql modules wasn't compiled. Did you check
the requirements for mysql and read the compile errors? It's not shown
here.



 And the asterisk console is flooded with these errors :

 [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707
 determine_firstline_parts: Bad request protocol Packet
 [Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707
 determine_firstline_parts: Bad request protocol Packet

 So for the next time to come i'll turn back to 1.2 :)

The chan_sip messages was only warnings, nothing serious. Propably  
strange NAT Keepalives, like those
I've seen from cirpak devices. Communication should work as expected.

If you give up for these errors, you might consider buying Asterisk  
Business Edition
where everything is precompiled and easy-to-install, and you have  
support.

Thanks for the feedback!

Best regards,
/Olle

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Dovid B]

Dave,
I agree with you. I think it would be smarter to go to a new format how ever 
one issues that a lot of people seem to have is when the syntax is changed. 
This is why I suggested both. Maybe there can be a month (or maybe even two) 
long discussion between the users and dev list for A) Current formatting B) 
formatting for the future and we can have both say for the next two major 
releases (as opposed to 1 now) and then move over. Wouldn't this make more 
people happy ?


- Original Message - 
From: dave cantera [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, December 20, 2007 6:33 AM
Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!


 dovid...
 while this seems like a good idea to have both sip show channels and
 show channels sip having two, three or even four ways to do the same
 thing would confuse/cripple the learning curve... * would turn into a
 microsoft mentality where there are dozens of ways to
 configure/reconfigure some of their products...  word, for example, can
 be configured with or without the tool bars and then you can configure
 hot-keys...  in fact, you can configure some products so that someone
 who learns it with a hacked config, could not possibly use the original
 stock config...  sorry to go on about this but it is one of my hot
 buttons...
 daveC

 Dovid B wrote:
 - Original Message - 
 From: Steve Edwards [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, December 19, 2007 5:43 AM
 Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's 
 old!



 On Sat, 15 Dec 2007, Johansson Olle E wrote:


 I wonder if there are any major obstacles for upgrading.

 How about the change from a bad command line interface to a really bad
 command line interface?

 I mean, Seriously? (in a Grey's Anatomy kind of way...)

 The old syntax was inconsistent -- show manager command vs sip show
 channels and just plain bad -- for example sip reload should have 
 been
 reload sip.

 The new syntax continues down the noun-verb path instead of correcting
 itself and using verb-noun like most other applications (MySQL, GDB,
 Oracle, etc.)

 Then, just to make it worse, now I have to learn which commands somebody
 (arbitrarily) decided are core and which are not -- for what benefit?
 Certainly doesn't make MY job easier!

 Approach the command line like a noob. I want Asterisk to show me
 something so I'll start the command line with show. I'm not quite sure
 what I'm doing, so I'll press TAB to see what I can show. Oh, 
 channel
 looks like what I want. Hmm, too much. Maybe I should have qualified 
 what
 kind of channel I'm looking for BEFORE the word channel.

 Here's a suggestion -- stop thinking like a parser and start thinking 
 like
 a person :)

 Which makes more sense (at least in English)?

  1) show black dogs -- show sip channels
  2) black show dogs -- sip show channels
  3) dogs black show -- channels sip show
  4) show dogs black -- show channels sip
  5) black dogs show -- sip channels show
  6) dogs show black -- channels show sip

 Is it too late to fix this for 1.6?

 Thanks in advance,


 I think as many people have pointed out they are used to a lot of 
 commands
 out there so changing it yet again would make more people unhappy. But 
 maybe
 asterisk can have both. Why not sip show channels for the old timers and
 show channels sip or show sip channels for the n00b's. Why shouldn't
 asterisk have both options ?



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 -- 
 My wife's sister is in California.
 I should buy her a Videophone2008!

 Truly, The Next Best Thing to Being There!
 --

 WorldWideVideoPhones.com
 856.380.0894




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Re: [asterisk-users] Upgrade to Asterisk 1.4 - testing

[Johansson Olle E]

20 dec 2007 kl. 01.43 skrev Dovid B:

 snip
 Our problem is that very few in the community test beta releases
 or development code. I want to send a big thank you to all that do,
 you are very important in this process. And for those of you who
 want to join, go to www.asterisk.org and find instructions on how
 to download development code for testing. Join the whoever
 tests this stuff group today :-)
 /snip

 Olle I would love to test but I do not know what I am looking for. I  
 would
 say that I have a fairly good knowledge of Asterisk  however I am  
 not the
 best at tracing the root problems of issues. I have no problem of  
 loading
 the bleeding edge version on a spate box, loading my current configs  
 on it
 and seeing where it goes down. Maybe some info on what to look for  
 when
 there are issues would help.

Dovid,
For people that wants to help the process, there's always time and
a large attention span from the development team. Join the #asterisk-dev
channel on IRC freenode.net and you'll find a weird enivronment (many
jokes among friends) but also a lot of people that can help you get  
going,
give you ideas for testing and respond to your ideas. There's usually
a lot of real-time activity there (US time, not on my mornings here in
Sweden at GMT+1), but it might slow down now for Xmas.

Welcome!

/O

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Tzafrir Cohen]

Hi

On Wed, Dec 19, 2007 at 12:19:08AM -0500, dave cantera wrote:
 ok, here is my $0.02...  I created a script since I had to 
 install/update so often and for various reasons...
 you can choose to compile automatically or manually...
 modify the current release numbers, your repository, and source root... 
 all else is automated..
 which is unloading zap driver, stopping a running asterisk, getting the 
 current release, untar'ng it and compiling it...
 enjoy,
 daveC

You can find my take on the subject at
http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/
I improved the existing scripts from bristuff to be more potent, as
explained in
http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html

The bristuff scripts have a little wrapper install.sh that calls
download.sh (downloads and patches. Kind of like rpmbuild -bp) and
compile.sh (builds and installs).

That separation can reduce some of the need for user interaction in your
script.

If you want to use them, I figure you should just remove the patching
commands and then you should be able to use those scripts mostly
unchanged.

 
 
 #!/bin/sh
 #
 #get_latest_rel.sh
 #
 # Dave Cantera:  [EMAIL PROTECTED]
 #
 #get the current asterisk release components, put them in our REPOSITORY
 #and unpack them in SRC_ROOT
 
 --- Change to suite between these lines --
 VER_AST=1.4.16
 VER_ZAPTEL=1.4.7.1
 VER_LIBPRI=1.4.3
 VER_ADDONS=1.4.5
 
 REPOSITORY=/root/tarballs
 SRC_ROOT=/usr/local/src
 --- Change to suite between these lines --
 
 HTTP_SITE=http://downloads.digium.com;
 PUB_DIR=/pub
 
 TARBALL_AST=/asterisk/releases/asterisk-${VER_AST}.tar.gz
 TARBALL_LIBPRI=/libpri/releases/libpri-${VER_LIBPRI}.tar.gz
 TARBALL_ZAPTEL=/zaptel/releases/zaptel-${VER_ZAPTEL}.tar.gz
 TARBALL_ADDONS=/asterisk/releases/asterisk-addons-${VER_ADDONS}.tar.gz
 
 COMPONENTS=${HTTP_SITE}${PUB_DIR}${TARBALL_AST}
 ${HTTP_SITE}${PUB_DIR}${TARBALL_ZAPTEL}
 ${HTTP_SITE}${PUB_DIR}${TARBALL_LIBPRI}
 ${HTTP_SITE}${PUB_DIR}${TARBALL_ADDONS} 
 
 echo
 echo
 echo  we are prepared to get the complete current release 
 echo  of asterisk, libpri, zaptel, and addons 
 echo  the tarballs will be placed in our REPOSITORY and 
 echo  then extracted to our SRC_ROOT 
 echo
 echo --- Activity Recap 
 echo
 echo  TARBALL REPOSITORY: ${REPOSITORY}
 echoSRC_ROOT: ${SRC_ROOT}
 echoasterisk tarball: ${TARBALL_AST}
 echo  libpri tarball: ${TARBALL_LIBPRI}
 echo  zaptel tarball: ${TARBALL_ZAPTEL}
 echo  addons tarball: ${TARBALL_ADDONS}
 echo
 echo -n  Are You Ready?  Y to procced: 
 read ANSWER
 
 if [ null${ANSWER} == nullY ]

# a matter of style:
case $ANSWER in Y* | y*) :;; 
  *) echo  Aborted by user ;;
  exit 0
esac

# and good bye to unneeded nesting.

 then
 echo
 echo -
 echo  stopping asterisk 
 echo
 echo  choose your poison: 
 echo  a) /usr/bin/asterisk -xr stop now
 echo  b) /etc/init.d/asterisk stop 
 echo
 echo -n   which one? 
 read STOPCMD
 if [ null${STOPCMD} == nulla ]
 then
 /usr/bin/asterisk -r -x 'stop now'
 fi
 if [ null${STOPCMD} == nullb ]
 then
 /etc/init.d/asterisk stop
 fi
 
 echo
 echo -
 echo  get the current asterisk  component releases and put them in 
 our repository ${REPOSITORY}
 # lets go to the repository directory
 cd ${REPOSITORY}
 
 for TARBALL in `echo ${COMPONENTS}`
 do
 echo getting component: ${TARBALL} 
 #wget ${TARBALL}

Err... one needs to uncomment that line, I guess.

I tend to like using 'wget -c' . Otherwise strange things may happen if
I press ctrl-C in the middle of the download.

Sadly, the current downloads.digium.com will make you re-download the
tarballs 

 done

 TARFILES=
 asterisk-${VER_AST}.tar.gz
 libpri-${VER_LIBPRI}.tar.gz
 zaptel-${VER_ZAPTEL}.tar.gz
 asterisk-addons-${VER_ADDONS}.tar.gz 

 echo
 echo -
 echo  unpack the current asterisk  component tarballs into our 
 source root ${SRC_ROOT}
 # lets go to the source root directory
 cd ${SRC_ROOT}
 for TARBALL in `echo ${TARFILES}`
 do
 echo untar'ng component: ${TARBALL} 
 #tar xzf ${TARBALL}
 done

 echo
 echo -
 echo  unloading Zap drivers
 # unload the zaptel drivers
 ZAP_MODULES=`lsmod | grep zap | awk '{printf(%s,,$4)}' | sed 's/,/ 
 /g'`

 for MODULE in `echo ${ZAP_MODULES}`
 do
 echo unloading zap module: ${MODULE}
 #modprobe -r ${MODULE}
 done
 
 echo
 echo  now you are ready to compile at ${SRC_ROOT} 
 echo
 
 echo -n  Shall I continue with the compile? Y?
 read COMPILE
 if [ null${COMPILE} == nullY ]
 then
 echo  Compiling Zaptel 

Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[dave cantera]

tzafrir,
thanks for the note. btw, Great docs!
asciidocs looks cool too!
thanks!
daveC

Tzafrir Cohen wrote:

  Hi

On Wed, Dec 19, 2007 at 12:19:08AM -0500, dave cantera wrote:
  
  
ok, here is my $0.02...  I created a script since I had to 
install/update so often and for various reasons...
you can choose to compile automatically or manually...
modify the current release numbers, your repository, and source root... 
all else is automated..
which is unloading zap driver, stopping a running asterisk, getting the 
current release, untar'ng it and compiling it...
enjoy,
daveC

  
  
You can find my take on the subject at
http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/
I improved the existing scripts from bristuff to be more potent, as
explained in
http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html

The bristuff scripts have a little wrapper install.sh that calls
download.sh (downloads and patches. Kind of like rpmbuild -bp) and
compile.sh (builds and installs).

That separation can reduce some of the need for user interaction in your
script.

If you want to use them, I figure you should just remove the patching
commands and then you should be able to use those scripts mostly
unchanged.

  
  

#!/bin/sh
#
#get_latest_rel.sh
#
# Dave Cantera:  [EMAIL PROTECTED]
#
#get the current asterisk release components, put them in our REPOSITORY
#and unpack them in SRC_ROOT

--- Change to suite between these lines --
VER_AST="1.4.16"
VER_ZAPTEL="1.4.7.1"
VER_LIBPRI="1.4.3"
VER_ADDONS="1.4.5"

REPOSITORY="/root/tarballs"
SRC_ROOT="/usr/local/src"
--- Change to suite between these lines --

HTTP_SITE="http://downloads.digium.com"
PUB_DIR="/pub"

TARBALL_AST="/asterisk/releases/asterisk-${VER_AST}.tar.gz"
TARBALL_LIBPRI="/libpri/releases/libpri-${VER_LIBPRI}.tar.gz"
TARBALL_ZAPTEL="/zaptel/releases/zaptel-${VER_ZAPTEL}.tar.gz"
TARBALL_ADDONS="/asterisk/releases/asterisk-addons-${VER_ADDONS}.tar.gz"

COMPONENTS="${HTTP_SITE}${PUB_DIR}${TARBALL_AST}
${HTTP_SITE}${PUB_DIR}${TARBALL_ZAPTEL}
${HTTP_SITE}${PUB_DIR}${TARBALL_LIBPRI}
${HTTP_SITE}${PUB_DIR}${TARBALL_ADDONS} "

echo
echo
echo " we are prepared to get the complete current release "
echo " of asterisk, libpri, zaptel, and addons "
echo " the tarballs will be placed in our REPOSITORY and "
echo " then extracted to our SRC_ROOT "
echo
echo "--- Activity Recap "
echo
echo " TARBALL REPOSITORY: ${REPOSITORY}"
echo "   SRC_ROOT: ${SRC_ROOT}"
echo "   asterisk tarball: ${TARBALL_AST}"
echo " libpri tarball: ${TARBALL_LIBPRI}"
echo " zaptel tarball: ${TARBALL_ZAPTEL}"
echo " addons tarball: ${TARBALL_ADDONS}"
echo
echo -n " Are You Ready?  Y to procced: "
read ANSWER

if [ "null${ANSWER}" == "nullY" ]

  
  
# a matter of style:
case "$ANSWER" in Y* | y*) :;; 
  *) echo " Aborted by user ";;
  exit 0
esac

# and good bye to unneeded nesting.

  
  
then
echo
echo "-"
echo " stopping asterisk "
echo
echo " choose your poison: "
echo " a) /usr/bin/asterisk -xr stop now"
echo " b) /etc/init.d/asterisk stop "
echo
echo -n "  which one? "
read STOPCMD
if [ "null${STOPCMD}" == "nulla" ]
then
/usr/bin/asterisk -r -x 'stop now'
fi
if [ "null${STOPCMD}" == "nullb" ]
then
/etc/init.d/asterisk stop
fi

echo
echo "-"
echo " get the current asterisk  component releases and put them in 
our repository ${REPOSITORY}"
# lets go to the repository directory
cd ${REPOSITORY}

for TARBALL in `echo ${COMPONENTS}`
do
echo "getting component: ${TARBALL} "
#wget ${TARBALL}

  
  
Err... one needs to uncomment that line, I guess.

I tend to like using 'wget -c' . Otherwise strange things may happen if
I press ctrl-C in the middle of the download.

Sadly, the current downloads.digium.com will make you re-download the
tarballs 

  
  
done
   
TARFILES="
asterisk-${VER_AST}.tar.gz
libpri-${VER_LIBPRI}.tar.gz
zaptel-${VER_ZAPTEL}.tar.gz
asterisk-addons-${VER_ADDONS}.tar.gz "
   
echo
echo "-"
echo " unpack the current asterisk  component tarballs into our 
source root ${SRC_ROOT}"
# lets go to the source root directory
cd ${SRC_ROOT}
for TARBALL in `echo ${TARFILES}`
do
echo "untar'ng component: ${TARBALL} "
#tar xzf ${TARBALL}
done
   
echo
echo "-"
echo " unloading Zap drivers"
# unload the zaptel drivers
ZAP_MODULES=`lsmod | grep zap | awk '{printf("%s,",$4)}' | sed 's/,/ 
/g'`
   
for MODULE in `echo ${ZAP_MODULES}`
do
echo "unloading zap module: ${MODULE}"
#modprobe -r ${MODULE}
done

echo
echo " now you are ready to compile at ${SRC_ROOT} "

Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Patrick]

Hi Steve,

On Tue, 2007-12-18 at 19:43 -0800, Steve Edwards wrote:

 The old syntax was inconsistent -- show manager command vs sip show 
 channels and just plain bad -- for example sip reload should have been 
 reload sip.

I agree. Reload sip would be the logical thing.

[snip]

 Approach the command line like a noob. I want Asterisk to show me 
 something so I'll start the command line with show. I'm not quite sure 
 what I'm doing, so I'll press TAB to see what I can show. Oh, channel 
 looks like what I want. Hmm, too much. Maybe I should have qualified what 
 kind of channel I'm looking for BEFORE the word channel.

That makes sense to me. It's also what I'm used to from working with
other equipment.

 Here's a suggestion -- stop thinking like a parser and start thinking like 
 a person :)
 
 Which makes more sense (at least in English)?
 
   1) show black dogs -- show sip channels
   2) black show dogs -- sip show channels
   3) dogs black show -- channels sip show
   4) show dogs black -- show channels sip
   5) black dogs show -- sip channels show
   6) dogs show black -- channels show sip

Excellent example. I'll put my 0.2 cents on #1 :) 

 Is it too late to fix this for 1.6?

I sincerely hope not. Your example shows that the CLI could use some
TLC. Let's hope the powers that be agree.

+1

Regards,
Patrick


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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Patrick]

Hi Olle,

On Wed, 2007-12-19 at 08:20 +0100, Johansson Olle E wrote:
[snip]
 The old way was a mess. We had two different systems, one like your
 old show  and one syntax starting with the module name. We had
 to move forward with only one syntax and decided to go for modulename  
 verb
 which is not human language-like, but we haven't really clamed that the
 CLI is a human language parser. Maybe we should go for an avatar
 approach...

I have not followed this discussion but the decision is quite puzzling
to me. Why would you make the human interface to Asterisk not human
language-like? That's just not logical. Were the devs expecting that the
majority of users would be HAL2000 clones instead of humans? :)

[snip]

 I do understand the pain with changing the CLI though, I hate to switch
 from Asterisk 1.0 to 1.2 to 1.4 and trunk and have different commands.

This is only an issue for developers and existing users who have (a
combination of) 1.0, 1.2 and 1.4 boxes and upgrade to a version with an
improved CLI. New users who get the latest major version of Asterisk
(assuming that version has the improved human language-like CLI) don't
have that issue. I don't mind the CLI differences because at some point
I move all my boxes to the new major release so only have to deal with
one version of the CLI at any time. Change usually means one needs to
adopt and an improved CLI seems worth it to me.

Regards,
Patrick


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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[James Collier]

I think it should be core dogs show black.

Seriously though, I think you make a good point.



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Steve
Edwards
Enviado el: miercoles, 19 de diciembre de 2007 4:43
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's
old!


On Sat, 15 Dec 2007, Johansson Olle E wrote:

 I wonder if there are any major obstacles for upgrading.

How about the change from a bad command line interface to a really bad 
command line interface?

I mean, Seriously? (in a Grey's Anatomy kind of way...)

The old syntax was inconsistent -- show manager command vs sip show 
channels and just plain bad -- for example sip reload should have been 
reload sip.

The new syntax continues down the noun-verb path instead of correcting 
itself and using verb-noun like most other applications (MySQL, GDB, 
Oracle, etc.)

Then, just to make it worse, now I have to learn which commands somebody 
(arbitrarily) decided are core and which are not -- for what benefit? 
Certainly doesn't make MY job easier!

Approach the command line like a noob. I want Asterisk to show me 
something so I'll start the command line with show. I'm not quite sure 
what I'm doing, so I'll press TAB to see what I can show. Oh, channel 
looks like what I want. Hmm, too much. Maybe I should have qualified what 
kind of channel I'm looking for BEFORE the word channel.

Here's a suggestion -- stop thinking like a parser and start thinking like 
a person :)

Which makes more sense (at least in English)?

1) show black dogs -- show sip channels
2) black show dogs -- sip show channels
3) dogs black show -- channels sip show
4) show dogs black -- show channels sip
5) black dogs show -- sip channels show
6) dogs show black -- channels show sip

Is it too late to fix this for 1.6?

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Tzafrir Cohen]

On Wed, Dec 19, 2007 at 02:40:21PM +0100, James Collier wrote:
 
 I think it should be core dogs show black.

You should use color instead of black to make the comparison more
valid.

  show dog color

Doesn't sound right (Here's a colour for you, doggy. Fetch!).

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

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Tilghman Lesher]

On Wednesday 19 December 2007 07:40:21 James Collier wrote:
 I think it should be core dogs show black.

No, that violates the pattern.  dogs is not a verb.  core show black dogs
or dogs show black would be the correct form.

-- 
Tilghman

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[MatsK]

Tilghman Lesher wrote:
 On Wednesday 19 December 2007 07:40:21 James Collier wrote:
   
 I think it should be core dogs show black.
 

 No, that violates the pattern.  dogs is not a verb.  core show black dogs
 or dogs show black would be the correct form.
   

Could this CLI syntax move over to the dev list, since it's mobing
further away from the original question!

/M
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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Patrick]

On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote:
 On Wednesday 19 December 2007 07:40:21 James Collier wrote:
  I think it should be core dogs show black.
 
 No, that violates the pattern.  dogs is not a verb.  core show black dogs
 or dogs show black would be the correct form.

Sorry but I'm not a native English speaker and I don't get it. Why is
dogs show black the correct form as opposed to the imho more correct
(in spoken language) show black dogs?

Regards,
Patrick



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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Steve Edwards]

On Wed, 19 Dec 2007, Patrick wrote:

 On Wed, 2007-12-19 at 08:33 -0600, Tilghman Lesher wrote:
 On Wednesday 19 December 2007 07:40:21 James Collier wrote:
 I think it should be core dogs show black.

 No, that violates the pattern.  dogs is not a verb.  core show black dogs
 or dogs show black would be the correct form.

 Sorry but I'm not a native English speaker and I don't get it. Why is
 dogs show black the correct form as opposed to the imho more correct
 (in spoken language) show black dogs?

It's not. I think it was a humorous reply to a humorous reply.

The core bit should die, die, die.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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