Asterisk 11 to 13 Upgrade Notes
You might be impacted by the upgrade to Asterisk 13 if you have:
- custom dialplan
- custom Asterisk configuration
- custom application using AGI, AMI or any other Asterisk interface
- custom application exploiting CEL or queue_log
- custom Asterisk modules (e.g. codec_g729a.so)
- customized Asterisk in some other way
- DAHDI trunks using SS7 signaling
If you find yourself in one of these cases, you should make sure that your customizations still work with Asterisk 13.
Some of the more common changes to look for:
- SS7 support is not available in the Asterisk package of XiVO between version 15.13 and 16.08 inclusively.
- All channel and global variable names are evaluated in a case-sensitive manner. In previous versions of Asterisk, variables created and evaluated in the dialplan were evaluated case-insensitively, but built-in variables and variable evaluation done internally within Asterisk was done case-sensitively.
- The SetMusicOnHold dialplan application was deprecated and has been
removed. Users of the application should use the CHANNEL function's
- The WaitMusicOnHold dialplan application was deprecated and has been
removed. Users of the application should use MusicOnHold with a
- The SIPPEER dialplan function no longer supports using a colon as a delimiter for parameters. The parameters for the function should be delimited using a comma.
- The SIPCHANINFO dialplan function was deprecated and has been removed. Users of the function should use the CHANNEL function instead.
For SIP, the codec preference order in an SDP during an offer is slightly different than previous releases. Prior to Asterisk 13, the preference order of codecs used to be:
- Our preferred codec
- Our configured codecs
- Any non-audio joint codecs
Now, in Asterisk 13, the preference order of codecs is:
- Our preferred codec
- Any joint codecs offered by the inbound offer
- All other codecs that are not the preferred codec and not a joint codec offered by the inbound offer
- Queue strategy
rrmemory(Round robin memory) now has a predictable order. Members will be called in the order that they are added to the queue. For agents, this means they will be called in the order they are logged.
- When performing queue pause/unpause on an interface without specifying an individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at least one member of any queue exists for that interface. This has an impact on the agent performance statistics; an agent must be a member of at least 1 queue for its pause time to show up in the statistics.
You can see the complete list of changes from the Asterisk website:
The AGI protocol did not change between Asterisk 11 and Asterisk 13; if you have custom AGI applications, you only need to make sure that the dialplan applications and functions you are using from the AGI are still valid.
List of known bugs and limitations for Asterisk 13 in XiVO:
When direct media is active and DTMF are sent using SIP INFO, DTMF are not working properly. It is also impossible to do an attended transfer from the Wazo Client in these conditions.