Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: SARK devs on July 01, 2009, 12:01:58 PM
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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.
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
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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:
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?
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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
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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
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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,
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OK old friend...
I had a play with it after dinner this evening (couldn't resist - I am such a dork :oops: ).
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
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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? :???:
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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
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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
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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
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
module load chan_sip
This will either work or, at least, tell you what the problem is.
Kind Regards
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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
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Oh my goodness!@##$@#
I'm starting to see head aches GLORE pouring out of the wood work here in Asterisk 1.6
I'm guessing They've change syntax left right & center therefore the autogeneration of .conf files needs a lot of updating before it will be compatible right?
I guess that's why you said (for the curious)
Cheers
-Matt
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Hello Mat
OK the three user files - as soon as you do a commit from Globals then the files will be created if they don't already exist. In earlier releases of asterisk it didn't matter if they weren't there. In 1.6 they have to be. Don't forget that SARK 2.3.2 is basically asterisk 1.4. For 1.6 proper we'll move this create code into the migrate/startup code.
Now the gotcha...
They changed the syntax rules in 1.6 (the arrogant !@##$@#) 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
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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
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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
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Sorry but I am a newbie to all of this so how exactly would I go about doing that?
Is this done from the server console or somewhere else and what is the command I should run?
I apologize for the questions but I have no telco experience and this is certainly a whole other experience but one I'm wanting to learn.
I tried setting up a couple SJphone v1.65.377a softphones and I got them to work but I get an error after a couple of seconds "SJphone Call terminated unexpectedly. Call signaling chanel closed unexpectedly." The connection says opperational despite the error but I can't seem to find why it is doing that. Also, is there a newer version of SJphone available since I believe they were bought by MagicJack.
-edb
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OK... please don't take this the wrong way, but if you are a newbie then we would strongly recommend that you run a more stable setup. SME8 and Asterisk 1.6 are both very early release products and you really need to know them both in some detail before you attempt to run them.
You can find out how to install a more stable set in our wiki...
http://sarkpbx.com/twiki
Kind Regards
S
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I took your advice and downloaded the "sark-sme-7.3-2.2.1-617.iso" and loaded it into a VM and everything is working fine now with the X-lite softphones.
Thank you for pointing this out to me and I am grateful to learn on a stable release.
Can you tell me if I should have the yum repos enabled and which ones so that nothing gets inadvertently screwed-up by an update?
Thanks again S
-edb
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Best not to run yum update unless you are comfortable with how asterisk kernel modules integrate with the kernel. There is a write up on the Wiki if you are interested.
Kind Regards
S
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The said file is not found - both in SAIL v2.3.2-19 and v2.4.1-2:
# dahdi_genconf
-bash: dahdi_genconf: command not found
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No - I'm afraid not. That file is part of Dahdi.
Kind Regards
S