If you are using a Linksys SPA942 and would like to enable viewing of other users status, or be able to press a button to call that extension here is how to configure Make sure you have the latest phone firmware 6.1.5(a). Connect to the web interface on the SPA942 and under the info tab.
I setup a US number with anveo $2.49 setup and $1.49 per month for a DID. Online chat support was helpful and they have a comprehensive call routing system including being able to accept faxes on the DID, setup voicemail etc. Call quality is good from Australia to US numbers. Incoming calls worked fine on.
Due to attempted hacks on VOIP boxes we need to increase security by allowing SIP access only from the relevant IPs using iptables firewall In the case below the Asterisk VOIP server is sitting behind a NAT firewall and has the relevant config set for this. I found upon enabling the linux firewall that while.
If running your asterisk server behind a firewall you may experience these symptoms: internal users can call mobile extensions configured with follow me and the call goes through and can be heard with no problem. If you have a rule configured so that external callers after a timeout calling internal extensions are then routed to.
After doing a backup of asterisk from one system, and restoring to a fresh build, you may find that when someone goes to voicemail, there is a seconds silence then it drops out. If you use *97 and try to record a message, it immediately drops back to the options to save the message without.
Below are a list of commands you can use from the Asterisk command line interface (CLI) The CLI is reached by using the Linux shell command asterisk-
The key folders to backup for asterisk are /etc/asterisk /var/log/asterisk /var/lib/asterisk /var/spool/asterisk /usr/lib/asterisk /var/lib/mysql /usr/local/tomcat/webapps You can use this script to back them up into one gz file. ora=`date ‘+%Y.%m.%d-%H.%M.%S’` tar zcf etc-asterisk-$ora.tar.gz /etc/asterisk tar zcf var-log-asterisk-$ora.tar.gz /var/log/asterisk tar zcf var-lib-asterisk-$ora.tar.gz /var/lib/asterisk tar zcf var-spool-asterisk-$ora.tar.gz /var/spool/asterisk tar zcf usr-lib-asterisk-$ora.tar.gz /usr/lib/asterisk tar zcf var-lib-mysql-$ora.tar.gz /var/lib/mysql tar.
I recently found this method of doing the backup/restore http://www2.elastix.org/en/component/kunena/116-security/47135-how-to-backup-elastix-for-migration.html?Itemid=58 A few things to be aware of Make sure the Elastix version is the same or higher than the version that you are transferring the backup .tar file from. Otherwise if you try to transfer to an earlier version the extensions will not transfer across.
When we put a call on hold using out asterisk server the Music on hold would chop in and out. We found this choppiness only occurred using Pennytel and Engin, other VOIP providers the MOH was smooth. Try this: Make sure you’ve compiled zaptel and have ztdummy loaded. Login to the server. If connecting from.
When opening the Call Detail Reports section of FreePBX, you may get the following error: YOU MUST ACCESS THE CDR THROUGH THE ASTERISK MANAGEMENT PORTAL! This occurs when the php session save path is not writable by the httpd process. 1. Determine the save path: grep save_path /etc/php.ini should result in something like: session.save_path =.