Asterisk 1.8 package

robi

oops, in directories section of asterisk.conf I see this:

astdbdir => /tmptmptmptmpvar/db/asterisk

and the directories required are not created in the filesystem (/var/log/asterisk/), nor the symlink, just checked.

marcelloc

Hi robi,

I've pushed menu fix and fixed preg_match for conf files

whait 15 minutes and reinstall package

https://github.com/bsdperimeter/pfsense-packages/commit/5e1c10abcc307efd9188959f867f509eb27b1107

Create an account at github so you can push patches to it too.

att,
Marcello Coutinho

mauricioniñoavella

marcelloc congratulations, by the contribution

thanks for this great work

marcelloc I'm glad that we have people like you

a hug and greeting friend

Mauricio

robi

@marcelloc:

Create an account at github so you can push patches to it too.

I'd love to. Is there a tutorial somewhere about this, as I'm not familiar about it at all.

marcelloc

@robi:

I'd love to. Is there a tutorial somewhere about this, as I'm not familiar about it at all.

try this micro how to for pfsense's github repo

http://forum.pfsense.org/index.php/topic,44686.msg232239.html#msg232239

mdima

Hi Marcello,
just for curiosity, how does this package work in a CARP environment? Ok I can point the LAN shared IP from the VOIP phones in the company (so only the Master box receives connections from the phone), but do both boxes try to registrate to the VOIP provider (the box currently working as Master and the one working as Slave)?

Thanks,
Michele

marcelloc

That's a good point.

In an outbound scenario it may work with carp as clients will reauth with asterisk.

For inbound calls, you can test configuring asterisk to listening on carp ips and see if backup asterisk will not crash.

mdima

@marcelloc:

That's a good point.
In an outbound scenario it may work with carp as clients will reauth with asterisk.
For inbound calls, you can test configuring asterisk to listening on carp ips and see if backup asterisk will not crash.

well, I am also worry about Asterisk try to register to the VOIP provider, then the VOIP provider will try to contact both boxes for an incoming call… I don't know if I can test that in the real environment, I will coordinate with my colleague that follows the telephony services in my company and try to imagine how we can manage a try. Now we use Freeswitch on a server in our DMZ network...

marcelloc

In theory backup box will not be able to register as it will not have the configured ip on it.

robi

@marcelloc:

@robi:

I'd love to. Is there a tutorial somewhere about this, as I'm not familiar about it at all.

try this micro how to for pfsense's github repo

http://forum.pfsense.org/index.php/topic,44686.msg232239.html#msg232239

I made changes, comitted, named "Updated asterisk package to remove errors in the log, cosmetic GUI fixes". The question is, when will they appear when reinstalling the package?

marcelloc

Did you did a pull request for this?

there is no alerts on https://github.com/bsdperimeter/pfsense-packages notifications

robi

Sorry please try now.

robi

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" ??? ??? ???

marcelloc

You need to wait the commit or cancel your current pull request and push another

robi

In this case please pull the first one in, I'll do some tests, and afterwards I'll push next ones if needed following.

marcelloc

Ok, I'll do it sunday.

I'm on smartphone now.

robi

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.

marcelloc

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 dialplans

I 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.

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.

asterisk-gui.png_thumb

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…