Asterisk 1.8 package
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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.
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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.
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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 -
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.
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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
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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
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any idea if FreePBX will be able to be compatible with Asterisk 1.8 on PFS ?
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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.
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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.
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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….
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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 -
http://downloads.asterisk.org/pub/telephony/asterisk-gui
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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.
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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 -
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/scriptshi, 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/imageupdatewhat is the command line here????
Thanks.
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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.
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Great Job.
But I found the app_meetme.so is not installed in the asterisk.
this module is very important for us!
Can you install it in the asterisk system?
Maybe you should comiple the dahdi module first!
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No dahdi for now :(
The dahdi port for freebsd is not working.
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not working?
http://svn.digium.com/svn/dahdi/freebsd/trunk/
I have compiled it successfully in my freebsd 8.1 and copied them to pfsense 2.0
see the following information.
[2.0-RELEASE][root@pfSense.localdomain]/etc/asterisk(30): kldstat
Id Refs Address Size Name
1 39 0xc0400000 ea8378 kernel
2 13 0xc6d25000 36000 dahdi.ko
3 2 0xc6e4e000 e000 dahdi_voicebus.ko
4 1 0xc6f97000 5000 dahdi_dynamic.ko
5 1 0xc6f9f000 2000 dahdi_echocan_jpah.ko
6 1 0xc6fa5000 3000 dahdi_echocan_kb1.ko
7 1 0xc6fa9000 3000 dahdi_echocan_mg2.ko
8 1 0xc6fae000 2000 dahdi_echocan_sec.ko
9 1 0xc6fb4000 c000 wcb4xxp.ko
10 1 0xc700d000 52000 wct4xxp.ko
11 1 0xc70bb000 e000 wctdm.ko
12 1 0xc70cd000 5000 wcfxo.ko
13 1 0xc70d2000 8000 wcte11xp.ko
14 1 0xc70df000 b000 wcte12xp.ko
15 1 0xc72ca000 7000 joskm.ko
I rebuilded the asterisk 1.8.x in pfsense,included the app_meetme.so.
conference now is running…...
But I do not have some FXS or FXO or E1/T1 cards,So i do not know whether the drived can run in the pfsense system. -
not working?
http://svn.digium.com/svn/dahdi/freebsd/trunk/
I have compiled it successfully in my freebsd 8.1 and copied them to pfsense 2.0
see the following information.Using ports? I need this way to build the package.
===> Building package for dahdi-2.4.0rc5_5 tar: include/dahdi/compat/types.h: Cannot stat: No such file or directory tar: include/dahdi/user.h: Cannot stat: No such file or directory tar: include/dahdi/wctdm_user.h: Cannot stat: No such file or directory tar: Error exit delayed from previous errors. pkg_create: make_dist: tar command failed with code 256 *** Error code 1 make package ===> dahdi-kmod-2.4.0rc5_4 may not be packaged: Should be in sync with the kernel to work correctly.If so, can you share the steps you did?
