Asterisk 1.8 package
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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
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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
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Manager (AMI): Disabled
Web Manager (AMI/HTTP): Disabled
Call data records: Enabled
Realtime Architecture (ARA): Disabled -
Directories
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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 ?
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To save logs to the USB stick I think it would be enough to just change the logdir path in asterisk.conf```
astlogdir => /var/log/asteriskFor a remote syslog server you can try some tips found on the internet: http://www.savelono.com/linux/configuring-asterisk-for-a-remote-syslog-server.html -
Over here on pfSense-2.1-RELEASE-4g-amd64-nanobsd_vga after package installation console outputs:
Parse error: syntax error, unexpected $end in /usr/local/pkg/asterisk.inc on line 389
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HEllo ,
In 2.1-RELEASE (i386)
built on Wed Sep 11 18:16:22 EDT 2013
FreeBSD 8.3-RELEASE-p11With clean package install i am wondering to activate the AEL module , but there is an error.
nxtwall*CLI> module load pbx_ael.so
Unable to load module pbx_ael.so
Command 'module load pbx_ael.so ' failed.
[Feb 10 11:18:17] WARNING[92895]: loader.c:481 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/pbi/asterisk-i386/lib/asterisk/modules/pbx_ael.so: Undefined symbol "destroy_pval"
[Feb 10 11:18:17] WARNING[92895]: loader.c:894 load_resource: Module 'pbx_ael.so' could not be loaded.Thanks for your help.
Regards
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I never used pbx_ael.so module. I guess it depends on something, which is missing. Perhaps googling on the error message could give some hints…
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I just upgraded my pFsense to the latest 2.1.2 release….
After the reboot & successfull update pFsense gave me the following message "Packages are currently being reinstalled in the background."Nice !
I have 2 packages....a small one (CRON) and Asterisk.
The problem is the size of the new Asterisk package that is automatically installed !!!!!- previous release - asterisk-1.8.25.0-i386.pbi -> 41MB
- current release - asterisk-1.8.26.1-i386.pbi -> 108MB
At the end of the installation Asterisk burns 580MB of space which leaves my 1GB CF card....voiceless...as my VoIP installation.
In fact even with a 2GB CF card it has been problematic for me to install it.
88% space used & installation goes forever. Then reboot, install the package again (in fact it will just run the installation phase not the download) and finally....it works ! But my card is nearly full with 88% of 907M used.Is there a reason for such a size increase ?
What would be the way for installing the older release instead ?Thanks for your help.
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Hmm…
I also noticed that on my 4G CF card the space usage was higher after the upgrade, but I thougt that was because I was also upgrading architecture from i386 to x64. Your post shows me that I was most probably wrong.
As far as I can remember the older Asterisk had some security issues, so the upgraded version is recommended to be used. However it's indeed surprising more than twice disk space to be used.
The older asterisk had quite a lot of non-functional modules, because of many missing dependencies. Maybe here these were fixed, that would explain the increased disk space requirerements.I don't know.
However, I suggest to move to 4GB card on NanoBSD...
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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 have the same behaviour here and using your workaround does the trick.
But it is annoying…. -
Hello ,
I am using Asterisk package in the new Pcengines APU board with Pfsense 2.1 .
I am searching to use Codec G729 ? there is someone who try it?Regards
Daniel
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I think i need to recompile the sources from here http://asterisk.hosting.lv/
in a FreeBSD 8.0 system , with the linux compatibility ( Pfsense 2.1 doenst)Tell me if i am in the right way !!
Thanks for your help
Regards
Daniel
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Hi All,
Does somebody tried the odbc connector from pfsense to another host?
I'm trying to configure it (to use a remote mysql server as dialplan backend) but can't configure properly the odbc on pfsense.
After adding the driver and the DSN, with isql I receive segmentation fault.
It is a missconfiguration or bug? -
I have a IPSEC VPN connections between 2 locations.
Location 1 Has a pfsense router (10.0.1.254) with a dedicated Asterisk server (10.0.1.2) behind the routers LAN port.
Location 2 is a pfSense Netgate router with Asterisk installed on the router (10.0.9.254)
At location 2 anything behind the Netgate router LAN port (10.0.9.254) ipsec traffic travels to Location 1 (10.0.1.0/24) fine with out issues. But from inside the Netgate router (10.0.9.254) either via trying to ping in the GUI or ssh the Location 1's network (10.0.1.0/24) no packetes travel down the VPN.
So my issues is that I have short codes (ie *80) to dial the two locations but since Asterisk is not using the VPN tunnel on the Location 2 Netgate router (10.0.9.254) they time out.
I need to find out how to get Asterisk from Location 2 to communicate down the VPN. Right now it simply does not even see the 10.0.1.0/24 network at location 1 from inside the router.
Any idea how to get inner office extension to work over vpn? My other locatios with dedicated Asterisk servers behind the router/firewall work fine. I can not get any location with asterisk as an add on to register as a sip peer over the vpn
