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Alex Rudd Segmentation fault after registering and loading g729a codec on x86-64 clean install
I am new to Asterisk and have purchased a g729 codec from Digium for use with a clean install of Asterisk. I am using FreePBX-64bit-6.12.65.iso Distro on a Dell T110 server. Everything installs and runs correctly.
I have followed the steps in the g729 README:
1) Download and execute the 'register' utility to generate a valid license.
2) Download and execute the 'benchg729' utility to determine the optimum build.
3) Use the 'G.729 Selector' web utility to determine the recommended G.729 codec binary download package (barcelona_64)
4) Download and install the 'codec_g729' binary that is built for your platform.
I have extracted the codec and copied it to the /usr/lib64/asterisk/modules/ directory. After doing this, I am loading the codec with the following command:
[root@localhost ~]# asterisk -rx "module load codec_g729a.so"
When I then try to enter the asterisk CLI, I am getting the following error:
[root@localhost ~]# asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
Rebooting the server brings up a looping error refering to /usr/sbin/safe_asterisk: line 158: 17898 Segementaiton fault (core dumped) and a scrolling text.
Can someone suggest what I can do to troubleshoot this further please, and also if this might be related to trying to use a 64-bit version of FreePBX rather than the 32-bit version?
thanks!
Best Answer chosen by Brian (Digium) Brian (Digium) This is not possible. The reason being is that every extension then can setup their phonebooks however they want. There is no global Switchboard phonebook that can be loaded for every extension. If you would like to see this feature as a future software feature, please send your request to features@switchvox.com.
Larry Prikockis Switchvox memory leak -> High swap usage since upgrade to 5.8.5
Alternatively, is there a way to downgrade a Switchvox appliance to an earlier version of the software?
Best Answer chosen by Larry PrikockisNicholas (Sangoma) There is a resolution slated for 5.9.1 which should be out very shortly.
If you have support, you can open a tech support ticket and we can install a script that runs a cron job every hour to keep the memory down until the formal release of the fix.
If you would like to rollback the version, you will need to have a backup made on the earlier version as backups must be restored to the same version they were taken on.
You would lose any configurations, voicemail, logs, that were done after that backup.
Here is a link to steps on restoring:
http://support.digium.com/articles/FAQ/How-to-rollback-to-a-previous-Switchvox-version?retURL=%2Fapex%2FKnowledge&popup=false
Thank you,
Scott McAlonan How do I adjust the page a Rapid Dial contact shows up on?
We are on version 6.2.1 if that is important.
Best Answer chosen by Scott McAlonanMalcolm DavenportWayne Stevens Can I prioritize call queues in 6.2? I would like to give certain queues priority over other queues.
Best Answer chosen by Wayne StevensNicholas (Sangoma) You cannot give queues priority over other queues unless calls simply hit them first, then rollover to the 2nd queue.
You can give priority to calls within a queue in 6.2.1 (look at the 'Call-Control Options for Queues' section in the link below:
http://kb.digium.com/articles/FAQ/Updating-to-Switchvox-6-2
Paul Zimmerle Conference phone can't dial external numbers
Is there a way to prevent this so that the phone is able to add external lines while in a call? I don't want to prevent it from dialing extensions.
Best Answer chosen by Paul ZimmerleMatthew HiltonFor the ones using Digium Feature Packs, there are a few things you can try...
Go to > Setup > Phone Feature Packs > Modify Phone Setup Options.
If you're using the default Dial plan (x.T|*xx.T), with the default Timeouts of 3|3...
You could try changing the timeout values to something longer, like 5 seconds, but I'm doubtful that will help.
i'd try changing the Digit Map to something more robust, such as this: [2-9]11|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxxxxT|[1-9]xxT
That will give you all the *11 dials (411,911, etc.), plus 11,10, and 7 digit dialing, and finall extension dialing. If you use this dial plan, you'll need to update the timeout settings to match, so,
3|3|3|3|3
Reboot the phone(s) after saving to get the new settings and give it a whirl. If that works - then you can do the same for the manually configured phones, but you'll have to do it in the config file or web gui for the phone itself.
Problem fixed: I was following the README files provided by email when I purchased my license without thinking them through properly. While I had correctly identified that I needed to modify the wget commands for the x86-64 version of the software I was using, I had not modified the link for the version of Asterisk I was using from 1.8.4 to 11.0 therefore
wget http://downloads.digium.com/pub/telephony/codec_g729/\asterisk-1.8.4/x86-64/codec_g729a-1.8.4_3.1.7-barcelona_64.tar.gz should actually have been wget http://downloads.digium.com/pub/telephony/codec_g729/\asterisk-11.0/x86-64/codec_g729a-11.0_3.1.7-barcelona_64.tar.gz for me (note the path to the different Asterisk version).
Once I corrected this, the codec loaded correctly and license shows on my server.