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| author | Kevin Harwell <kharwell@digium.com> | 2017-06-13 14:17:29 -0500 |
|---|---|---|
| committer | Kevin Harwell <kharwell@digium.com> | 2017-06-13 14:28:11 -0500 |
| commit | 3a8cb4986d1d7dfb73248c042782e0aea5c2a2c1 (patch) | |
| tree | 0b4ab91f1df42a03c15a0d743c0ad73ea820e4a4 /res/res_pjsip_session.exports.in | |
| parent | 7994914c882874593eb17fdd8c910ee418f44a1b (diff) | |
res_pjsip_refer/session: Calls dropped during transfer
When doing an attended transfer it's possible for the transferer, after
receiving an accepted response from Asterisk, to send a BYE to Asterisk,
which can then be processed before Asterisk has time to start and/or
complete the transfer process. This of course causes the transfer to not
complete successfully, thus dropping the call.
This patch makes it so any BYEs received from the transferer, after the REFER,
that initiate a session end are deferred until the transfer is complete. This
allows the channel that would have otherwise been hung up by Asterisk to
remain available throughout the transfer process.
ASTERISK-27053 #close
Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a
Diffstat (limited to 'res/res_pjsip_session.exports.in')
| -rw-r--r-- | res/res_pjsip_session.exports.in | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/res/res_pjsip_session.exports.in b/res/res_pjsip_session.exports.in index a39485e66..fdfc5fb47 100644 --- a/res/res_pjsip_session.exports.in +++ b/res/res_pjsip_session.exports.in @@ -3,6 +3,7 @@ LINKER_SYMBOL_PREFIXast_sip_session_terminate; LINKER_SYMBOL_PREFIXast_sip_session_defer_termination; LINKER_SYMBOL_PREFIXast_sip_session_defer_termination_cancel; + LINKER_SYMBOL_PREFIXast_sip_session_end_if_deferred; LINKER_SYMBOL_PREFIXast_sip_session_register_sdp_handler; LINKER_SYMBOL_PREFIXast_sip_session_unregister_sdp_handler; LINKER_SYMBOL_PREFIXast_sip_session_register_supplement; |