Upgrading to Asterisk-1.6.0.1 & FreePBX-2.5.1.0

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So I'm looking at tying Elastix in with Microsoft OCS 2007 for a client, looks like I need SIP over TCP which is only in Asterisk-1.6.

Time to try the update I thought!
I'd previously compiled my own 1.4.22 without a worry, I needed iLBC, and have built a few distributions from a debian net-inst so thought this couldnt be too hard :) Figured I'd write a quick How-To just to show that its not too difficult, and also so I can just copy / paste everything next time I want to do an upgrade.

This assumes you're clever enough to locate all the files & download them yourself (Well I'd really like to hope so. If you cant - You shouldnt be upgrading)!!!
Files are:
asterisk-1.6.0.1.tar.gz
dahdi-linux-2.0.0.tar.gz
dahdi-tools-2.0.0.tar.gz
freepbx-2.5.1.tar.gz

Maybe: asterisk-addons-1.6.0.tar.gz

First, login as root, then put all files in /usr/src
tar xvzf dahdi-linux-2.0.0.tar.gz
cd dahdi-linux-2.0.0
make
make install
# Note: I dont have my TDM410p in this box, so I havent actually tested the calling in / out using any hardware cards etc...
# It'll download a whole lot of firmware and stuff afterwards, just leave it to do its thing. When its done, it will tell you that you should now install the dahdi-tools package.
# We're doing everything in this order, so that Asterisk builds with dahdi support.
cd ..
tar xvzf dahdi-tools-2.0.0.tar.gz
cd dahdi-tools-2.0.0
./configure
make
make install
# You may also want to run make config - it'll copy some sample config files which may or may not help you
cd ..
tar xvzf asterisk-1.6.0.1.tar.gz
cd asterisk-1.6.0.1
# run the following if you want iLBC - Skip it otherwise
./contrib/scripts/get_ilbc_source.sh
./configure
# When in the menuselect, go into Codec Translators, scroll down to codec_ilbc and hit Enter to enable it, then press F12 to save & exit
make menuselect
make
# It'll take a few mins to build. It didnt take too long on my 1.6Ghz Atom
make install
# Note, it'll complain afterwards about all the modules that arent compatible, things like g729, write them down somewhere
cd /usr/lib/asterisk/modules

# Now you mv all the files it complained about to somewhere else, otherwise it quite possibly wont start. Unfortunately this means you may lose SQL CDR logging amongst other things

# Once you've done that, we just need to install FreePBX-2.5.1!
cd /usr/src
tar xvzf freepbx-2.5.1.tar.gz
cd freepbx-2.5.1
./install_amp
amportal restart


Around now, it failed due to missing format_au.so which apparently is Sun Microsystems AU format (signed linear).
I commented it out by adding a ; in front of the following line in /etc/asterisk/modules.conf:
load => format_au.so

Ran:
amportal restart
Now its as good as gold :)
So it kinda sucks Ive lost SQL CDR logging for now (Cant be bothered messing around re-enabling anything - its almost midnight here in New Zealand), and Ive only run a few test calls, nothing major, but its still nicely integrated with Elastix and hasnt died on me as the 1.6 beta versions did. This isnt a high volume box, mostly just used for testing purposes, so Ive not really put it through its paces. This at least shows its relatively easy to upgrade to the new asterisk-1.6.0.1 & freepbx-2.5.1 :)

Comments / feedback welcome, as well as thoughts on how to stop it crashing when trying to load cdr_addon_mysql.so, as this is why I move the modules all to /root ;)

Cheers


Chill.
 
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Haha!
Searching google for information on Asterisk 1.6 and SIP over TCP, I find this thread only 10 hours after the initial post.

Good to know that Google (appears to) index the forums here regularly! :)
 
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Have enabled TCP (I think - Getting a softphone to test now) by editing /etc/asterisk/sip_general_custom.conf & adding:
tcpenable=yes

Was dead simple!
 
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Cool Chill. B)

Now, with your information, maybe the futur Elastix version can be made with Freepbx 2.5.1 and asterisk 1.6.0.1?! ;)
 
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The upgrade to Asterisk-1.6.0.1 was a bigger deal than the new FreePBX-2.5.1 to be honest.

Either way, Ive had no issues so far, and have been doing a fair bit of stress testing today :D
Tying it in with MS OCS 2007 looks just that little bit closer now ;)

Perhaps I should chuck this up on the Wiki...
 
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For me, i wait the SIP TCP and SIP TLS. ;)
We try to connect asterisk why the MOCS and Call Manager CISCO.
(With the Call Manager: no problem)
 
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Dear Chilling_Silence, after Upgrading to Latest Version FreePBX 2.5.1 have you no problem with the Elastix Embeded FreePBX on "extension options all fields are empty", i see only this problem after Upgrading to Latest Version FreePBX 2.5.1.
 
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Ive had no problems at all. Everything is still there, and everything integrates perfectly!

Ive just re-done everything as-above on a machine last night with perfect success! :)
It looks fine on both the embedded FreePBX & Un-embedded FreePBX.
I can take screenshots if you want? Ive noticed one or two minor differences with the new FreePBX version, new features...
 
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No problems, happy to help!

If only I could remember how to make RPM's and do testing in that way... I made some for Yoper Linux back in the day (Some 5-odd years ago now I think) but I'm pretty sure I've forgotten everything :p

BTW - Calling over TCP (instead of UDP) works brilliantly!

I bought the ZoIPer Biz version which supports TCP, and only forwarded TCP traffic at my Router (As well as specifying it in ZoIPer), so I connected from the Softphone -> Asterisk through TCP, then Asterisk was connected to my ITSP through UDP! Brilliant stuff!!

Testing TLS is next on the to-do list ;)
 
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Rafael,

Who do I see about contributing to Elastix? I cant specifically program, but I'm sure I can assist in other ways.. Flick me an email perhaps?
I tried to PM you but couldnt see a way to in these forums...
 
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Great, thanks a lot. On my signature is my email ;)

rbonifaz[at]elastix.org
 
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You have a more specific guide.

I tried with all versions of elastix to migrate to 2.6 already been a problem only remaining errors and most of the time remaining in bug and q calls are delayed in any quantity and exit.

dahdi not use even if not completely dahdi another package of Digium better.:(
 
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Errr... Do you mean Asterisk-1.6 or FreePBX-2.5?

Are you able to explain a little more the issues you've been having? Broken english?
 
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Hello friends,

Happy New Year

I have a doubt, I can upgrading only Asterisk to 1.6 and doesnt freePBX? It doesnt bring to me any problems?

Thanks
 
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Hello,

I have tried to upgrading to asterisk 1.6 and when ran amportal restart appears this:

STARTING ASTERISK
/usr/sbin/safe_asterisk: line 125: 19143 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 125: 19175 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed

-----------------------------------------------------
Asterisk could not start!
Use 'tail /var/log/asterisk/full' to find out why.
-----------------------------------------------------

I see the log, and appear this error:

[Jan 6 11:43:05] WARNING[19175] loader.c: Error loading module 'format_au.so': /usr/lib/asterisk/modules/format_au.so: cannot open shared object file: No such file or directory

"Around now, it failed due to missing format_au.so which apparently is Sun Microsystems AU format (signed linear).
I commented it out by adding a ; in front of the following line in /etc/asterisk/modules.conf:
load => format_au.so"

Now ran amportal restart and apperas this:
STARTING ASTERISK
/usr/sbin/safe_asterisk: line 125: 19143 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 125: 19175 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed

-----------------------------------------------------
Asterisk could not start!
Use 'tail /var/log/asterisk/full' to find out why.
-----------------------------------------------------

In the log appear this:
[Jan 6 11:48:02] VERBOSE[19297] logger.c: == Manager registered action AGI
[Jan 6 11:48:02] VERBOSE[19297] logger.c: == Registered application 'AGI'
[Jan 6 11:48:02] VERBOSE[19297] logger.c: res_agi.so => (Asterisk Gateway Interface (AGI))
[Jan 6 11:48:02] VERBOSE[19297] logger.c: == Registered file format wav, extension(s) wav
[Jan 6 11:48:02] VERBOSE[19297] logger.c: format_wav.so => (Microsoft WAV format (8000Hz Signed Linear))
[Jan 6 11:48:02] VERBOSE[19297] logger.c: == Registered file format pcm, extension(s) pcm|ulaw|ul|mu
[Jan 6 11:48:02] VERBOSE[19297] logger.c: == Registered file format alaw, extension(s) alaw|al
[Jan 6 11:48:02] VERBOSE[19297] logger.c: == Registered file format au, extension(s) au
[Jan 6 11:48:02] VERBOSE[19297] logger.c: == Registered file format g722, extension(s) g722
[Jan 6 11:48:02] VERBOSE[19297] logger.c: format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz)

Please help me.

Thanks
 
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Hello,

Sorry for my english...........but what do you mean with: "Now you mv all the files it complained about to somewhere else"

But I cant understand if are the tar.gz?

Thanks and my apoligize for cant understand
 
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Sep 24, 2008
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Hello chilling,

I compiled the files and move the .tar.gz and the problems continues. I cant start the asterisk

Please help me.

Thanks
 
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Sep 23, 2008
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C'mon mate, I know you didnt re-read the first post!!

amportal restart

Around now, it failed due to missing format_au.so which apparently is Sun Microsystems AU format (signed linear).
I commented it out by adding a ; in front of the following line in /etc/asterisk/modules.conf:
load => format_au.so
Please, read the whole thing ;) I can understand when things dont work then hearts start racing, adrenaline gets pumping, but thats the time when you *most* need to keep a level-head on you :)
 

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