Asterisk 1.8 package

robi

That sounds good. My approach was simplistic.

modules:
For example app_db is problematic on nanobsd, as it cannot be saved on a read-only filesystem. Iax is a good thing, but rarely used nowdays, as it's not properly route-able protocol. timing_pthread - I'll have to look at this, but I remember having dependency problems on pfSense install. Disabling non-used modules saves memory too.
I'm using it flawlessly with 2 SIP phones and 4 SIP registrar accounts like this and never had any sound or whatever problems so far.

sip.conf:
Has specific recommendations for working properly on pfSense. Plus the original sip.conf is huge and over commented, not speaking that each .conf file is installed twice - keeping only one fully commented is more than enough.

But having a full-blown asterisk-GUI indeed implies to solve all the dependency problems properly - and after all, renders the simple status GUI useless, as all the functions I wrote I suppose, are present in the big GUI.

I'd love that too - but requires lots of work, I guess…

marcelloc

The simple gui will continue as many people like doing their own dialplan for custom and/or small instalation.

I'll publish your changes as well try to fix nanobsd erros, but config options must be something easy to change.

Advoc8tr

Can someone fill me in on how to configure asterix once it is installed? The service is running and the status page shows up but can't work out where / how to configure settings (SIP servers, IPs etc..)

marcelloc

In current version, you need to edit asterisk files

status-> asterisk -> edit configuration

sip.conf # define trunks,extensions and sip settings
extensions.conf # define dialplan

I'm planning to include asterisk-gui on this package but I need more time to finish it.

robi

Marcel, here's a small update to the package. I'm attaching it here rather than sending it via GIT, as it seems problematic that way.

In the PHP part I added a couple of cosmetics, and in asterisk.inc I added a check-routine for nanobsd, for existance of asterisk log, and callog dir.

If you didn't merge my old GIT changes so far nevermind, just overwrite the files with these.

asterisk_pkg_upd_robi_1.zip.xls

pkwong

Just wondering.. what are the chances of 2 features..

1. A show Iax trunks feature (or iax peers)
2. The ability to route incoming traffic directly to a sip trunk that has been created.

The reason I'm asking is due to NAT and OPENVPN settings, getting calls to a gateway is near impossible using sip and nat traversal even through siproxd with stun enabled.

I'd love to be able to route an iax trunk to the openvpn interface, have it connect to the asterisk plugin on the pfsense and have a pfsense asterisk sip trunk to the gateway on the far end, thus allowing me to communicate with the gateway on the far end without all the nat traversal.

So basically, here's how it would look:

Public Asterisk Machine (Public IP)<–-------------->IAX Trunk<--------------> pfsense openvpn gateway<------>Asterisk Plugin on PFsense<----------->SIP TRUNK ON LAN interface / subnet to Gateway with private IP (192.168.x.x/24)

So basically calls can come in the IAX trunk over the VPN connection and be routed to the pfsense asterisk plugin and immediately piped to the GSM Gateway behind the pfsense on the LAN Gateway.

marcelloc

@pkwong:

Just wondering.. what are the chances of 2 features..

1. A show Iax trunks feature (or iax peers)
2. The ability to route incoming traffic directly to a sip trunk that has been created.

These features(and others) will be available when I have time to finish asteriskgui port to pfnse
http://forum.pfsense.org/index.php/topic,47210.msg250379.html#msg250379

robi

@pkwong:

I'd love to be able to route an iax trunk to the openvpn interface, have it connect to the asterisk plugin on the pfsense and have a pfsense asterisk sip trunk to the gateway on the far end, thus allowing me to communicate with the gateway on the far end without all the nat traversal.

So basically, here's how it would look:

Public Asterisk Machine (Public IP)<–-------------->IAX Trunk<--------------> pfsense openvpn gateway<------>Asterisk Plugin on PFsense<----------->SIP TRUNK ON LAN interface / subnet to Gateway with private IP (192.168.x.x/24)

Don't you think this will dramatically increase audio delay in the phone lines? Especially the OpenVPN part of the road can add time delays to TCP/IP going thru the tunnel. Any delay more than 25ms/direction will make a phone conversation unusable as the dialogue will become unnatural.

EOC2611P

@robi:

Don't you think this will dramatically increase audio delay in the phone lines? Especially the OpenVPN part of the road can add time delays to TCP/IP going thru the tunnel. Any delay more than 25ms/direction will make a phone conversation unusable as the dialogue will become unnatural.

My last Skype conversation was having a delay of almost 2 minutes :D

robi

@EOC2611P:

@robi:

Don't you think this will dramatically increase audio delay in the phone lines? Especially the OpenVPN part of the road can add time delays to TCP/IP going thru the tunnel. Any delay more than 25ms/direction will make a phone conversation unusable as the dialogue will become unnatural.

My last Skype conversation was having a delay of almost 2 minutes :D

:D :D :D :D

npereira

any idea if FreePBX will be able to be compatible with Asterisk 1.8 on PFS ?

marcelloc

@npereira:

any idea if FreePBX will be able to be compatible with Asterisk 1.8 on PFS ?

Not yet. Freepbx is much more complex(and powerfull) then asterisk gui.

I've looked for freepbx integration but the setup involves a lot of extra steps and php5.3 that may break pfsense 2.0.1 install.

serialdie

@marcelloc:

@npereira:

any idea if FreePBX will be able to be compatible with Asterisk 1.8 on PFS ?

Not yet. Freepbx is much more complex(and powerfull) then asterisk gui.

I've looked for freepbx integration but the setup involves a lot of extra steps and php5.3 that may break pfsense 2.0.1 install.

freepbx plains sucks. is a pain in the ass to maintain on the back end.

torontob

Any delay more than 25ms/direction will make a phone conversation unusable as the dialogue will become unnatural.

You missed a 0. It's 250ms. But again it only starts to degrade slowly from there. I running at 400ms just fine here.

Yes, FreePBX is the worst option. Probably a nice simple css/mxl/html page would do for most of the features. Maybe copy FreePBX layout and features but not it's packages. They will never work with pfSense team to have something out that would be working for everything. Too politically charged….

xbaha

@marcelloc:

Good news!

I could get asterisk-gui 2.0 running on pfsense. ;D

Next step is adjust some gui options to have a full funcional asterisk package for pfsense.

Status from robi, gui configuration from digium, compilation and joining by me.

where can i find digium gui???
thanks

marcelloc

@xbaha:

http://downloads.asterisk.org/pub/telephony/asterisk-gui

xbaha

@marcelloc:

@xbaha:

http://downloads.asterisk.org/pub/telephony/asterisk-gui

sorry for this stupid question… i downloaded the package, unzip it, tried ./configure , something was missing, and also "make;make install" is unknown command, do i need to install something first?
sorry, but my Linux is basic knowledge.. pls check attached.
thanks for any help.

make.jpg_thumb

marcelloc

I did not finished it yet, but here are few steps you need after copying extracted files to the folder

mkdir /var/lib
mkdir /etc/dahdi/
mkdir /var/lib
mkdir /etc/dahdi/

ln -s /usr/local/etc/asterisk /etc/asterisk
ln -s /usr/local/share/asterisk /var/lib/asterisk

/var/lib/asterisk/gui_backups
/var/lib/asterisk/sounds/imageupdate

chown -R asterisk /var/lib/asterisk
chown -R asterisk /usr/local/etc/asterisk
chown -R asterisk /etc/dahdi

/var/lib/asterisk/static_html/config
/var/lib/asterisk/scripts

xbaha

@marcelloc:

I did not finished it yet, but here are few steps you need after copying extracted files to the folder

mkdir /var/lib
mkdir /etc/dahdi/
mkdir /var/lib
mkdir /etc/dahdi/

ln -s /usr/local/etc/asterisk /etc/asterisk
ln -s /usr/local/share/asterisk /var/lib/asterisk

/var/lib/asterisk/gui_backups
/var/lib/asterisk/sounds/imageupdate

chown -R asterisk /var/lib/asterisk
chown -R asterisk /usr/local/etc/asterisk
chown -R asterisk /etc/dahdi

/var/lib/asterisk/static_html/config
/var/lib/asterisk/scripts

hi, where exactly i should copy the extracted files? i don't need to compile them??
also, your code :


/var/lib/asterisk/gui_backups
/var/lib/asterisk/sounds/imageupdate

what is the command line here????

Thanks.

marcelloc

@xbaha:

what is the command line here????

maybe just folders to create. I'm not sure you need to compile it, just extract the files.

If I have time these days I'll try to publish a new package version with asterisk-gui.