[asterisk-dev] Asterisk 13.36.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.36.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.36.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29011 - chan_sip: ToHost property not cleared on
      reload
      (Reported by Dennis)
 * ASTERISK-28987 - BridgeCreated ARI event shows wrong
      video_mode info
      (Reported by sungtae kim)
 * ASTERISK-28927 - Asterisk crash in music on hold
     
      (Reported by David Cunningham)
 * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer
      triggered INVITE when NAT is active (UDP transport with
      external_media_address)
      (Reported by Michael Neuhauser)
 * ASTERISK-28995 - res_pjsip_registrar: Expires on statically
      configured contacts is not correct
      (Reported by tootai)
 * ASTERISK-28978 - acl: named_acl rule misconfiguration results
      in segfault on reading rule from realtime
      (Reported by
      Andrew Yager)
 * ASTERISK-28975 - res_http_websocket: Text payload data
      doesn't necessary include trailing zero
      (Reported by
      Nickolay V. Shmyrev)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.36.0-rc1

Thank you for your continued support of Asterisk!
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