Asterisk 1.8 package
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You need to compile it on freebsd 8.3
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How cool… One could use Asterisk to place directly mobile calls through a supported 3G modem.
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How cool… One could use Asterisk to place directly mobile calls through a supported 3G modem.
Yes. I've tested it on linux and it works. The problem I had way with usb power supply for multiple 3g modem and power boot sequence some times.
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Hi Marcelloc,
First, thanks for your great work again…
I was running a FreePBX server behind Pfsense, however it seems there are some NAT problem and I can't find any solution for that.
Then I tried your Asterisk package, it works perfect :D. The only question is it's difficult for me to edit the configurations... :-[
I read your post and you have installed Asterisk-gui successful before, I am now sure it's still working for latest pfsense 2.1?
Can I ask you to post a guide here?Thanks
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I've not tested it on 2.1 yet and freebsd port is still pending.
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Hi Marcelloc,
First, thanks for your great work again…
It's mainly a robi work. I just did some small modifications ;)
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The only question is it's difficult for me to edit the configurations… :-[
[/quote]if it is only to modify the config file directly, you can do it this way (if you use windows) :
- enable SSH in "System" -> "Advanced" - > "Admin Access" and check "Enable Secure Shell"
- use WINCSP to access the files in /conf/asterisk/
that's the way i modify my conf files for asterisk.
I hope the GUI will be ported soon. :) Will be good to have a complete GUI to manage configuration.
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Why don't you use edit configuration tab? ???
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I am running the Asterisk 1.8 package on pfsense 2.1 and have the following issues:
Everytime I reboot pfsense the following (demo) configuration text is added to sip.conf
==>
[301]
;demo extension for pfSense
type=friend
defaultuser=301
insecure=port,invite
secret=1234
regexten=301
host=dynamic
context=default[302]
;demo extension for pfSense
type=friend
defaultuser=302
insecure=port,invite
secret=1234
regexten=302
host=dynamic
context=default<==
Everytime I reboot pfsense my call history is lost
Everytime I reboot pfsense my Asterisk log file is empty
Everytime I reboot pfsense my personal voicemail message is goneAre there any sollutions to my above problems
Thanks in advance
Kr,
Matthijs
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What pfsense version are you using?
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Like I said I am using pfsense 2.1 :)
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O sorry
2.1-RELEASE (i386)
built on Wed Sep 11 18:16:44 EDT 2013
FreeBSD 8.3-RELEASE-p11 -
PBX Core settings
–---------------
Version: 1.8.19.0
Build Options: LOADABLE_MODULES
Maximum calls: Not set
Maximum open file handles: Not set
Verbosity: 0
Debug level: 0
Maximum load average: 0.000000
Startup time: 21:41:43
Last reload time: 21:41:43
System: FreeBSD/8.3-RELEASE-p3 built by root on i386 2012-12-19 14:50:46 UTC
System name:
Entity ID: xxx
Default language: en
Language prefix: Enabled
User name and group: /
Executable includes: Disabled
Transcode via SLIN: Enabled
Internal timing: Enabled
Transmit silence during rec: Disabled
Generic PLC: Enabled-
Subsystems
-------------
Manager (AMI): Disabled
Web Manager (AMI/HTTP): Disabled
Call data records: Enabled
Realtime Architecture (ARA): Disabled -
Directories
-------------
Configuration file:
Configuration directory: /conf/asterisk
Module directory: /usr/pbi/asterisk-i386/lib/asterisk/modules
Spool directory: /var/spool/asterisk
Log directory: /var/log/asterisk
Run/Sockets directory: /var/run/asterisk
PID file: /var/run/asterisk/asterisk.pid
VarLib directory: /usr/pbi/asterisk-i386/share/asterisk
Data directory: /usr/pbi/asterisk-i386/share/asterisk
ASTDB: /var/db/asterisk/astdb
IAX2 Keys directory: /usr/pbi/asterisk-i386/share/asterisk/keys
AGI Scripts directory: /usr/pbi/asterisk-i386/share/asterisk/agi-bin
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any news or suggestions ? ::)
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Not sure about voicemail but I keep logs and cdr after reboot. Still using 301/302 but no changes to them and 303+ externals are fine, no re-addition of demo extensions.
Working nicely on fresh 2.1 with raid1 (shouldn't matter). I have 2 lans on LAN and OPT1 and set localnet for both to pfsense's IP on those nets. Also doing 1:1 on WAN VIP to OPT2 so thinking of trying to get SIP to pass through tot that on a theoretical setup where pfsense handles some lans and passes an external IP to another lan that has it's own asterisk for whatever reason or just one phone talking to another outside asterisk or provider that has to be on 5060 as well.
Is there any explanation as to why jitterbuffer is on for PFSENSE's asterisk? I get why allowguest isn't… just curious about jitterbuffer.
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Well I finaly figured it out
In /usr/local/pkg
there is the a file called asterisk.inc
It contains the following code wich is causing my issue of the demo extentions being added after a reboot==>
if (strpos($script_file,'demo extension for pfSense') === false) { //first check if already added…
$add_demo_extension = << <eof<br>[301]
;demo extension for pfSense
type=friend
defaultuser=301
insecure=port,invite
secret=1234
regexten=301
host=dynamic
context=default[302]
;demo extension for pfSense
type=friend
defaultuser=302
insecure=port,invite
secret=1234
regexten=302
host=dynamic
context=defaultEOF;
$script_file .= $add_demo_extension;
file_put_contents($script, $script_file, LOCK_EX);
}
<==Please fix this in the next release
Thanks in advanceKr
:) :) :)</eof<br>
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As a workaround, all you have to do is keep the comment line "demo extension for pfSense" present in the conf file, but delete the demos themselves. The script only checks for the string "demo extension for pfSense", if it's present, it doesn't add anything.
I can't seem to reproduce this problem in my setup though. If I delete the whole thing, it keeps it deleted after reboot too.
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Hello,
I have a specific question concerning logging as my Asterisk seems to often miss the connection to my VoIP provider.
How could I send the Asterisk logging to a syslog server (I have one running in my house) ?Another solution could be to send the logs to a USB stick. But how ?
Tx for your help !
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no one ?
