Topic: Asterisk 1.6 with sail 2.3.1-20 on 8.0b4 (for the curious)

[SARK devs]

Here is a full asterisk 1.6.1  install from sources onto 8.0b4 with sail 2.3.1-20.   DO NOT put this onto a production machine.  It's just for testing and for the curious. - If it breaks anything, then you get to keep all the bits.

Code: [Select]

yum install gcc gcc-c++ ncurses-devel kernel-devel

cd /usr/src
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz
wget http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.6.0.tar.gz
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.2.0+2.2.0.tar.gz
tar xzvf asterisk-1.6.1.1.tar.gz
tar xzvf dahdi-linux-complete-2.2.0+2.2.0.tar.gz
tar xzvf asterisk-addons-1.6.0.tar.gz   
ln -s /usr/src/kernels/2.6.18-128.1.14.el5 linux
cd /usr/src/dahdi-linux-complete-2.2.0+2.2.0
make all
make install
make config
cd /usr/src/asterisk-1.6.1.1
./configure
make menuselect
make; make install
make samples
make config
cd /usr/src/asterisk-addons-1.6.0
./configure
make
wget http://sarkpbx.com/sail/sail-2.3.1/sail-2.3.1-20.noarch.rpm
yum localinstall sail-2.3.1-20.noarch.rpm --enablerepo=base
/etc/init.d/sark stop
/etc/init.d/sark start
/etc/init.d/sark stop

reboot

yum remove gcc gcc-c++ ncurses-devel kernel-devel

Best

S

[Franco]

Hi S,
I took a shot on this setup this weekend and succeed
I went for asterisk 1.6.2 which has the support for the MFCR2 I talked about in another thread!
Now correct me if I'm wrong, but it seems the inner parts of asterisk 1.6.x are very different from previous versions. I'm saying that due to errors like these:

Code: [Select]

The application delimiter is now the comma, not the pipe.  Did you forget to convert your dialplan?  (Dial(IAX2/voipsark@sarkvoip/7142||))
I'm using sail-2.3.2-11.
Will I have to go back to 1.4 to regain such precious functionalities?

[SARK devs]

Well, you may have to in the very short term..

We will replace pipes with commas in a near release.  Quite frankly, the Digium devs p*#% me off.  In 1.2 it was commas moving to pipes.  In 1.4 it was pipes and now in 1.6.2 they've gone back to commas again.   Backward compatibility?  Not at Digium. 

Rant over

Best

S

[SARK devs]

just run a couple of quick greps on the source code...

It looks like a couple of hours work to convert the generator and AGI back to comma delimiters.

Probably next weekend but not promising...



Best

S

[Franco]

Well, now that I have it down on the MFCR2 stuff, I´ll take a shot and see if I can have it running on 1.4.
If playing with the code is something that may help you, you can count on me. Email me directions and I´ll be glad to help.

Best,

[SARK devs]

OK old friend...

I had a play with it after dinner this evening (couldn't resist - I am such a dork  ).

I cut you an experimental rpm....

wget http://sarkpbx.com/sail/sail-experimentals/sail-2.3.2.12e-1.noarch.rpm

Seems to work OK on my rel 8.0 VM here.  Let me know if you hit any walls.  After the fix, the only pipe delimiters I could grep are in custom apps, which we can sort later. 

I will port it to a 7.4 VM tomorrow and check if it works there also.

Best

Jeff

[Uber_CaT]

ok a little off topic as I'm not use SME 8 beta
I've loaded a vanilla SME 7.4
then used your example above to try load sail / sark / asterisk
I modified the versions for the latest (non beta or RC's)
got thru the whole thing seemingly without error but my init.d has got no sark in it for me to stop or start
and the web interface has got none of the SARK menu's that I'd expect to be there?

here is a log of my installation as I did it:

Installed SME 7.4

yum update -y
signal event-upgrade
signal event-reboot

##From here I start to follow your guides more or less
yum install gcc gcc-c++ ncurses-devel kernel-devel
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.5.tar.gz
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addons-1.6.1.1.tar.gz
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz

##I always download this as well
wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-sounds-1.2.1.tar.gz

tar xzvf asterisk-1.6.1.5.tar.gz
tar xzvf asterisk-addons-1.6.1.1.tar.gz
tar xzvf dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz
tar xzvf asterisk-sounds-1.2.1.tar.gz

##here you can see my kernel version
ln -s /usr/src/kernels/2.6.9-89.0.9.EL-i686/ linux

cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0
make all
make install
make config

cd /usr/src/asterisk-1.6.1.5
./configure
make menuselect
##In the menu I only added a few more audio files for other codecs
make; make install
make samples
make config

cd /usr/src/asterisk-addons-1.6.1.1
./configure
make
##You didn't say run "make install" above but I ran it anyway.
make install

cd /usr/src
wget http://sarkpbx.com/sail/sail-2.3.2/sail-2.3.2-15.noarch.rpm
yum localinstall sail-2.3.2-15.noarch.rpm --enablerepo=base

##So now when I tried to run /etc/init.d/sark stop or start
##There is no sark inside init.d or like I meantioned no web interface options either.
##What did I do wrong?

[SARK devs]

HI there,

It all looks good...

I'm guessing that the yum localinstall for sail failed on a missing dependancy (although you don't show the output). 

To make sure do

rpm -q sail

I'm guessing it will tell you that sail isn't installed.

There is a dependancy in there that won't load automatically... Perl-File-ReadBackwards... you can install it by doing...

rpm -Uvh http://sarkpbx.com/sail/sail-2.3.2/perl-File-ReadBackwards-1.04-1.2.el4.rf.noarch.rpm

Then you can try your sail install again.  All the Asterisk stuff looks just fine from what I can see.  Easy to test it.. just type asterisk -vvvvc at the console and see if it comes up.  End it by typing "stop now".

Let us know how you get on

S

 

[Uber_CaT]

W00t! Perfect,
It was late obviously I wasn't paying close attention to the dependency failure.

I've installed TFTP & setup my DHCP with option 66 for auto provisioning
and that's working fine they get there extention settings,
but none my phones or softphones want to register

I have a horrible feeling I'm over looking something COMPLETELY obvious it's almost like the server isn't talking SIP I haven't installed ZAPTEL something I've always done in the past but this server doesn't have any hardware cards installed in it "yet" no analogue or PRI or ISDN.

SJphone says: "service unavailable Server not responding"
Snom 300 log says: "Registrar 6000@192.168.42.240 timed out"

I can't find a THING in asterisk logs or asterisk "CORE SET VERBOSE 10"
not a peep while trying to register the phone.

when in the asterisk prompt I can't do a "SIP SHOW PEERS" it's like SIP isn't there is that even possible?

Cheers
-Matt

[SARK devs]

I can't do a "SIP SHOW PEERS" it's like SIP isn't there is that even possible?

Yes it is possible that SIP isn't loaded.  At the Asterisk console you can do

Code: [Select]

core show channeltypes
This will show you which of the channel drivers are loaded.    If it isn't there (and it sounds like it won't be) you can attempt to load it with

Code: [Select]

module load chan_sip
This will either work or, at least, tell you what the problem is.

Kind Regards

[Uber_CaT]

Hi there,

ok well I noticed that during boot I get a
FATAL: Module dahdi not found

But I'm not too worried about that since I have no Hardware telephony devices yet.
(but shouldn't the module load but just fail loading the devices?)
(anyway this shouldn't effect sip i believe?)

"core show channeltypes"
only shows the following yes SIP seems to be missing:
Type        Description                              Devicestate  Indications  Transfer
----------  -----------                              -----------  -----------  --------
Skinny      Skinny Client Control Protocol (Skinny)  yes          yes          no
Agent       Call Agent Proxy Channel                 yes          yes          no
Console     OSS Console Channel Driver               no           yes          no
Local       Local Proxy Channel Driver               yes          yes          no
DAHDI       DAHDI Telephony Driver                   no           yes          no
USTM        UNISTIM Channel Driver                   no           yes          no
MGCP        Media Gateway Control Protocol (MGCP)    yes          yes          no
Phone       Standard Linux Telephony API Driver      no           yes          no

when trying to load sip using "module load chan_sip.so" this is what I get:

Unable to load module chan_sip.so
Command 'module load chan_sip.so ' failed.
SIP channel loading...
[Sep  8 17:18:31] ERROR[5569]: config.c:1083 process_text_line: The file 'sark_customer_sip_headers.conf' was listed as a #include but it does not exist.
[Sep  8 17:18:31] NOTICE[5569]: chan_sip.c:22520 reload_config: Unable to load config sip.conf

It seems that if an include file is missing it fails to load completely.
ok so I edited my sip.conf file and found 3 lines that I needed to comment out:
;#include sark_customer_sip_headers.conf
;#include sark_customer_sip_registrations.conf
;#include sark_customer_sip_devices.conf

but now I seem to be hacking this a little? should those file have been created?

Oh and I'm guess this has something too do with Asterisk 1.6, but it said this when I loaded SIP:
[Sep  8 17:24:00] NOTICE[5669]: chan_sip.c:22212 build_peer: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'

how this post isn't too messy and you can read it

Cheers
-Matt

[Uber_CaT]

Oh my goodness!@##$@#

I'm starting to see head aches GLORE pouring out of the wood work here in Asterisk 1.6

I'm guessing They've change syntax left right & center therefore the autogeneration of .conf files needs a lot of updating before it will be compatible right?

I guess that's why you said (for the curious)

Cheers
-Matt

[SARK devs]

Hello Mat

OK the three user files - as soon as you do a commit from Globals then  the files will be created if they don't already exist.    In earlier releases of asterisk it didn't matter if they weren't there.  In 1.6 they have to be.  Don't forget that SARK 2.3.2 is basically asterisk 1.4.  For 1.6 proper we'll move this create code into the migrate/startup code.

Now the gotcha...

They changed the syntax rules in 1.6 (the arrogant !@##$@#) and they aren't backward compatible.  However, SAIL-2.3.2-15 should generate syntactically correct 1.6 code.  If it doesn't then send us the errors and we'll fix 'em.

 
Best
 

[edb]

I have installed according to the instructions at the start of this thread on a new install of SME 8.0b4 and all appears to be working except when I go to the menu in Server Manager and go to PCI Cards I get the following error:
Software error:
Error occurred during DAHDI generation.

For help, please send mail to the webmaster (admin), giving this error message and the time and date of the error.
Now at this point and since I was just trying this in a VM so I do not have any PCI devices installed but is it necessary to have them installed for the menu to work? Just curious

-edb

[SARK devs]

No it is not necessary to have any devices defined.  The error occurs when SARK calls dahdi_genconf.  You can try running genconf manually from your console to see why it is complaining.

Best

S