| Commit message (Collapse) | Author | Age |
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1. for a given UA, each call has a unique line number starting
from 1
2. the list of calls is sorted in an arbitrary order, but the
last list element indicates "current call"
3. the '@' command takes a numeric argument which is the
line-number of the wanted call
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1. Added support for RTP timeout. The feature is disabled by default
and can be enabled with config "rtp_timeout N" where N is the number
of seconds of RTP inactivity. If this is detected, the call is
closed with a "special" SIP reason code of 701.
2. Added support for automatic re-connect in the menu module.
This can be enabled by setting the 2 config items:
redial_attempts 3
redial_delay 5
This work was contributed by Sveriges Radio. Thanks
goes to Ola Palm and Jim Eld.
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- The network instance is now in struct network and does not
use any local/static data
- A new top-level struct in baresip.c owns the single instance
of struct network
it is a long-term goal to remove all local/static data
from libbaresip and make it fully re-entrant.
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- added ua_hold_answer() which will first put on-hold the active
call (if exist) and then answer the new incoming call
fixes #50
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- change from a memory-reference to a const pointer
- this saves one mem_deref() in the destructor and also
decreases the chance of introducing memory leaks
- test on debian
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most of this work was done by Juha Heinanen, and reviewed by me.
thanks!
- menu: add support for sending SIP OPTIONS requests to any peer,
using the 'o' command
- presence: add support for PUBLISH, set ;pubint=N to activate
- ua: add Allow/Supported header to OPTIONS reply
add MESSAGE to allowed methods
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- metric_init calls tmr_start(). tmr_start must be called from the
re_main() thread.
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- sip_param_decode() renamed to msg_param_decode()
- use msg_ctype_cmp() to check for Content-Type
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