Asterisk 1.8 package
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Did you did a pull request for this?
there is no alerts on https://github.com/bsdperimeter/pfsense-packages notifications
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Sorry please try now.
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I made two more fixes, comitted etc. and when trying to send a new pull request it says "Oops! There's already a pull request for nagyrobi:master" ??? ??? ???
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You need to wait the commit or cancel your current pull request and push another
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In this case please pull the first one in, I'll do some tests, and afterwards I'll push next ones if needed following.
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Ok, I'll do it sunday.
I'm on smartphone now.
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I discovered further issues related to the package approach, but cannot go further until commits are pulled in (to see if I'm on the right track or not).
I need to learn about how daemon's logging works in pfSense - maybe we should use a similar approach in asterisk's case.
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Your pull request is marked as This pull request cannot be automatically merged..
I'm going to merge your changes by hand but module template and sip.conf forced is not a good idea. :(
for example:
res_timing_pthread.so is essential for audio quality(timing source) as dahdi is not installed.
chan_iax2.so is an excelent option for trunking and nat.
app_db.so is a very fast built in db for dialplansI know it´s not that simple but I think it's better trying to include a gui like asterisknow or freepbx instead of forcing configurations.
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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.
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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…
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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.
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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..)
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In current version, you need to edit asterisk files
status-> asterisk -> edit configuration
sip.conf # define trunks,extensions and sip settings
extensions.conf # define dialplanI'm planning to include asterisk-gui on this package but I need more time to finish it.
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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 ?
