I generally run my Asterisk boxes in a very limited virtual environment, so keeping memory usage to a minimum is definitely a good thing. I’ve seen some success in that department by scrapping Apache in place of nginx.
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Polycom BLF on Asterisk (and FreePBX)
There seems to be a lot of misinformation floating around on this subject, but after ignoring most of it and playing around I have a FreePBX system with BLF, remote caller ID, and directed pickup all working fine on my Polycom phone. Here’s how I did it, for the record.
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Asterisk voicemail notifications with MP3
I found a lovely script for sending voicemail notifications in MP3 format instead of wav. It’s a bit of a hack, since Asterisk only provides you with an entire, MIME encoded email message as input. Anyway, I improved it somewhat and wanted to share. This version adds call info and an image to the MP3 file, and is more elegant IMHO – e.g. use of pushd/popd or mktemp.
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Sending Asterisk 1.8 voicemails to multiple emails
Here is the difficult way to send Asterisk 1.8 voicemail notifications to more than one address. The easy way is to define a sendmail alias on your system. However, if you want to allow management of emails through a GUI like FreePBX, that won’t do much good for you.
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Sending Asterisk 1.4 voicemails to multiple emails
Here’s the complicated way to get Asterisk sending voicemail notifications to multiple email addresses. (The easy way? Use group mailboxes and/or sendmail aliases.) Once this modified module is compiled and installed, multiple addresses in voicemail.conf can be separated by a pipe character.
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A2Billing authentication of forwarded calls
So let me guess. You’re using A2Billing on FreePBX – I don’t know how TrixBox or AsteriskNOW work, but the concept is the same – and whenever you forward a phone to an outside number it doesn’t work. Callers get prompted with “Please enter the complete PIN number” instead of having the call completed.
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FreePBX hangs at login
Have you ever clicked on your FreePBX link, get prompted for a password, and then sit staring at your browser “connecting to…” your server forever? Restarting Apache doesn’t work, restarting MySQL doesn’t work, even restarting Asterisk doesn’t work. Meanwhile everything else is running just fine, but you need to restart the entire server before FreePBX starts working again.
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Configure a Cisco 7960 phone for use with Asterisk
This one’s sure to get some hits from Google.
First, if you need to upgrade the phone’s firmware, follow the directions on this page at Cisco. The phones come from Cisco configured to use SCCP instead of SIP, and you’ll almost always want to change the firmware. If you know what you’re doing, you’ll have that done in an hour. I read all these posts from people getting confused and agonizing over the firmware upgrade; 90% of their problems could have been avoided by following that document to the letter.
Anyway, once the phone’s firmware is upgraded you should do a factory reset on the phone. To do that, reboot it by pushing * + Settings + 6 together. As soon as the reboot starts, hold down the # key. After a second it will come up with a message about the factory reset key being detected. Type 123456789*0# and it will ask if you want to keep network settings. Push 2 for no.
Now your phone is wiped out. Most of the configuration files are detailed extensively elsewhere, but the important ones are SIPDefault.cnf and SIP<mac address>.cnf where <mac address> is of course the MAC address of your phone in uppercase. Here’s what you can get away with as a bare minimum (note that these directives can be put in either file, as far as I know):
image_version: "P0S3-07-4-00" ; your firmware image file
proxy1_address: "asterisk.example.com" ; you can put an ip address in as well
proxy_register: 1 ; you can't take incoming calls if you aren't registered
nat_enable: 1 ; obviously don't use this if you're not behind NAT
nat_received_processing: 1 ; same as the last one
telnet_level: 2 ; to get access to all functions in the telnet shell
line1_name: "username" ; your asterisk username from sip.conf
line1_authname: "username" ; this will be the same 99% of the time
line1_password: "secret" ; also from sip.conf
user_info: "none" ; I'm not 100% certain this is necessary
Something important to note is that these values get saved in the phone. That’s why we did the reset. If you don’t explicitly set a value to be empty (“”) it will use the value already in the phone, not the default value.
I spent a day getting “401 Unauthorized” messages back from Asterisk before I added those 2 NAT settings. I’m not sure what the problem was, because Asterisk was getting registration requests from the phone, but for some reason it was responding with 401 until I changes settings on the phone.
I’ll add for the record that getting into the phone via telnet can be very useful. The default password is “cisco” and debug sip-messages followed by register 1 1 is your friend. Also, erase protflash will remove all the setting stored in flash memory, forcing the phone to reget the settings from TFTP. Much better than doing a reboot to reload the config files.