Topic: [CONTRIB UPDATE] smeserver-trixbox.fws_beta4

[Daniel B.]

Hi everyone. I'm proud to announce the latest trixbox contrib for our favourite distro.
This package is still called smeserver-trixbox, even if now there's a lot of changes from trixbox.

The core of the package stay the same (asterisk, freepbx, fop, recordings), but the other stuff change:
- vtigerCRM 5.03 definitly replace sugarCRM (it's a patched version of vtiger which works with php4 and offers asterisk integration like click-to-dial and popup on incomming calls)
- meetme, maint, a2billing were removed as it represent a lot of work and I don't use it.
- hylafax + smeserver-hylafax (thanks dungog) + iaxmodem can be installed and configured, this way you can send faxs through one PSTN line, and without any additionnal physical modem
- All the rpms are downloaded directly from trixbox/dag/dungog repo
- zaptel modules are now rpm packaged (I just recompile the srpm from trixbox to match our kernel, only 2.6.9-55.0.2.EL and 2.6.9-55.0.6.EL both smp and non-smp are supported, if you're using an older kernel, please upgrade your server or the script won't install anything)
- The panel in the server-manager have changed, now there's a new section called trixbox


A lot of little bug fix and enhancement.

If you want to try:
http://sme.firewall-services.com/spip.php?rubrique5

[svangool]

Hi VIP-ire,

Thanks, I've been waiting for that!

Questions:
- Do you think that an update from your beta3 (Freepbx upgraded to 2.2.3 and Asterisk to 1.2.23) would work?
- What versions of Trixbox, Freepbx, Asterisk and Zaptel did you package?

Sjef

[Daniel B.]

Hi.

An upgrade from beta3... I'm not sure, I haven't try. One of the problem is that I've changed the WEBROOT in amportal.conf, now freepbx lives in /opt/asterisk/admin instead of /opt/asterisk/html/admin. But you can upgrade more or less manually. Just edit the trixbox script, and look at the begining, search where the repo are configured, you can set them manually, and upgrade all the rpms with yum --enablerepo=trixbox update. Then, to upgrade freepbx, look a the directory webapps in the archive, you'll find a directory called freepbx in which the archive of the latest freepbx is. Just extract this archive in your actual freepbx location and run the magic commande

Code: [Select]

./start_asterisk start
./install_amp
Of course you should backup everything before, and if you have a test server, you should try on it first.
For the versions of the rpms, it's the latest from trixbox as they are downloaded from trixbox repo (and some deps from dag and hylafax from dungog)
-asterisk-1.2.22-1
-zaptel-1.2.19-1
-freepbx 2.3
-vtigerCRM 5.03
- hylafax 4.3.4-1

[jmbac]

I attempted to install your contrib, the script is unable to complete installation with following error..  running on smeserver 7.2

SETTING FILE PERMISSIONS
Permissions OK
-------------------------------------------
Setting language in configuration files
-------------------------------------------
trixbox: line 443: install_panels: command not found
-------------------------------------------
Setting asterisk_cleanup to be executed daily
-------------------------------------------
-------------------------------------------
Setting Startup Scripts
-------------------------------------------



trixbox: line 484: syntax error near unexpected token `done'
trixbox: line 484: `    done'

Thanks
john

[Daniel B.]

Hi. I've just tested to download it and it's working, so I think you had a problem during your downloading. Can you retry ?
I've also updated the archive as there was a problem detcting the supported kernel for version 2.6.9-55.0.2.EL(smp), it's corrected now.

[svangool]

Hi VIP-ire,

My system was updated this week to kernel 2.6.9-55.0.6.EL with yum, does that make a difference for this package?
It did for your beta3, because I had to perform your trick http://forums.contribs.org/index.php?topic=34264.msg153820#msg153820 again yesterday to get Zaptel working (it did work with the 2.6.9-55.0.2.EL) with SME 7.2.

Sjef.

[Daniel B.]

HI.
There's no more need to do this dirty trick for zaptel to work, now I recompile the srpms from trixbox so they match SME kernels. In the package, there's support for version 2.6-55.0.2.EL and 2.6-55.0.6.EL both smp and non-smp. I'll try to do so each time a new kernel is released, and as quickly as possible.

You can get them here:
http://sme.firewall-services.com/downloads/smeserver-trixbox/rpms/

[clnsupport]

Thanks for all your effort and for delivering me hours of pleasure playing with trixbox on my sme server. I’ve got a question regarding restoring the backup your script makes when I remove it. Basically when I uninstall beta 3 how do I restore my settings to beta 4?

[Daniel B.]

Hi.
As this is still beta release, I do not provide upgrade support in the script. Maybe I'll do when I consider the product stable. For now, upgrade must be done manually. The global steps are the following:

Set the trixbox repository (edit the trixbox script and look for db yum_repository, you'll find the command which will set it)

yum --enablerepo=trixbox update asterisk asterisk-sounds asterisk-addons zaptel

This will update the most important rpms.

Then install the zaptel kernel modules, you can find it here:
http://sme.firewall-services.com/downloads/smeserver-trixbox/rpms/

Just download them and yum localinstall the one you need.

After, you can download the latest freepbx package from their website and perform the upgrade manually (documentation on their website http://freepbx.org).

Once this is done, you can install the fax stuff with
sh trixbox install_fax

I do not provide real support for upgrades now because this kind of installations are quite often personalized and it's hard to write a step by step how-to

[Nullig]

I keep getting the following error:

"Could not reload FOP server"

I've uninstalled/reinstalled a couple of times, but still get the error.

Any ideas?

Nullig

[jmbac]

Daniels,
Thanks for the contrib. it's installed OK and asterisk work great.. and I was trying to test the pop ups on vtigercrm.. are they working for you.. I  can only get clicktocall working..  Is there anything else I have to setup other than  the asterisk extension as SIP/xxxx


Thanks

[Daniel B.]

I keep getting the following error:

"Could not reload FOP server"

I've uninstalled/reinstalled a couple of times, but still get the error.

Any ideas?

Nullig

Hi. This error is "normal". I mean, I cannot find a easy solution to remove it. It's not veryy important, it just means that when you apply a configuration with freePBX, it cannot restart it (but it write the configuration files for fop). This problem is because on trixbox, apache is running under user asterisk, so it has permission to start/stop fop (which also runs under asterisk). I've manage to remove most of the error messages by adjusting permissiosn on file but not that one. If you want to reload fop, you can do so in the panel "security" in the server-manager, or on the command line: amportal restart_fop.

Daniels,
Thanks for the contrib. it's installed OK and asterisk work great.. and I was trying to test the pop ups on vtigercrm.. are they working for you.. I  can only get clicktocall working..  Is there anything else I have to setup other than  the asterisk extension as SIP/xxxx


Thanks

Hi. Yes, they are working, but there's a little configuration. I haven't take the time to write it yet, so here's a quick how-to. Imagine you want a ring group to be rung on incoming calls, which contains extensions 100 101 102 and 103. The ring group number in this example is 600. Create the ring group in freepbx. Now, log into your server and look at the file /etc/asterisk/extension_additional.conf. You should find something like this:

Code: [Select]

[ext-group]
include => ext-group-custom
exten => 600,1,Macro(user-callerid,)
exten => 600,n,GotoIf($["foo${BLKVM_OVERRIDE}" = "foo"]?skipdb)
exten => 600,n,GotoIf($["${DB(${BLKVM_OVERRIDE})}" = "TRUE"]?skipov)
exten => 600,n(skipdb),Set(__NODEST=)
exten => 600,n,Set(__BLKVM_OVERRIDE=BLKVM/${EXTEN}/${CHANNEL})
exten => 600,n,Set(__BLKVM_BASE=${EXTEN})
exten => 600,n,Set(DB(${BLKVM_OVERRIDE})=TRUE)
exten => 600,n(skipov),Set(RRNODEST=${NODEST})
exten => 600,n(skipvmblk),Set(__NODEST=${EXTEN})
exten => 600,n,Set(RecordMethod=Group)
exten => 600,n,Macro(record-enable,100-101-102-103,${RecordMethod})
exten => 600,n,Set(RingGroupMethod=ringall)
exten => 600,n(DIALGRP),Macro(dial,20,${DIAL_OPTIONS},100-101-102-103)
exten => 600,n,Set(RingGroupMethod=)
exten => 600,n,GotoIf($["foo${RRNODEST}" != "foo"]?nodest)
exten => 600,n,Set(__NODEST=)
exten => 600,n,dbDel(${BLKVM_OVERRIDE})
exten => 600,n,Goto(app-blackhole,hangup,1)
exten => 600,n(nodest),Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST})

Just copy this part in /etc/asterisk/extension_custom.conf, and change it like this:

Code: [Select]

[custom-ring-group]
include => ext-group-custom
exten => 600,1,Macro(user-callerid,)
exten => 600,n,GotoIf($["foo${BLKVM_OVERRIDE}" = "foo"]?skipdb)
exten => 600,n,GotoIf($["${DB(${BLKVM_OVERRIDE})}" = "TRUE"]?skipov)
exten => 600,n(skipdb),Set(__NODEST=)
exten => 600,n,Set(__BLKVM_OVERRIDE=BLKVM/${EXTEN}/${CHANNEL})
exten => 600,n,Set(__BLKVM_BASE=${EXTEN})
exten => 600,n,Set(DB(${BLKVM_OVERRIDE})=TRUE)
exten => 600,n(skipov),Set(RRNODEST=${NODEST})
exten => 600,n(skipvmblk),Set(__NODEST=${EXTEN})
exten => 600,n,Set(RecordMethod=Group)
exten => 600,n,Macro(record-enable,100-101-102-103,${RecordMethod})
exten => 600,n,Set(RingGroupMethod=ringall)
exten => 600,n,AGI,ident.agi|100
exten => 600,n,AGI,ident.agi|101
exten => 600,n,AGI,ident.agi|102
exten => 600,n,AGI,ident.agi|103
exten => 600,n(DIALGRP),Macro(dial,20,${DIAL_OPTIONS},100-101-102-103)
exten => 600,n,Set(RingGroupMethod=)
exten => 600,n,GotoIf($["foo${RRNODEST}" != "foo"]?nodest)
exten => 600,n,Set(__NODEST=)
exten => 600,n,dbDel(${BLKVM_OVERRIDE})
exten => 600,n,Goto(app-blackhole,hangup,1)
exten => 600,n(nodest),Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST})
The script which will display the popup is ident.agi
Now you can remove the ring group you have created at the begining in freePBX, it was just to have an example. And, in your inbound routes, select the destination to be a custom app with these parameters:

custom-ring-group,600,1

Now apply the config, and try an inbound call, you should have a popup on the account associated with extensions 100 101 102 103.

Cheers

[Nullig]

@VIP_ire

I finally solved the problem by changing the following line in the /etc/passwd file from:

asterisk:x:454:454:Asterisk VoIP PBX:/var/lib/asterisk:/bin/false

to

asterisk:x:454:454:Asterisk VoIP PBX:/var/lib/asterisk:/bin/bash

That got rid of the error.

BTW, thanks so much for the install script - you've done a terrific job there.

Nullig

[jmbac]

[root@devsrv ~]# /sbin/e-smith/db accounts set $URL asterisk-url comment 'reserved url for asterisk'
/sbin/e-smith/db dbfile set key type [prop1 val1] [prop2 val2] ...

[Daniel B.]

[root@devsrv ~]# /sbin/e-smith/db accounts set $URL asterisk-url comment 'reserved url for asterisk'
/sbin/e-smith/db dbfile set key type [prop1 val1] [prop2 val2] ...

What does it means? Can you explain with some more details the error?