Skip to content
Snippets Groups Projects
Select Git revision
  • asterisk_rdkb
  • 0505-invite
  • fix_multiple_dns_lookup
  • dev_fxs_no_wb
  • 14600-wenpeng
  • fix_fallback
  • 14666_fxs_no_wideband_codec
  • fix_srv_records
  • wenpeng-registration-test
  • fix_deadlock_in_bridge_peer_functions
  • devel default protected
  • securesip_pjproject_part1
  • PJ_Example_Review
  • fix_internal_calls_auth
  • fix_internal_calls
  • Secure_SIPtelephony_part1
  • SIPtelephony_DECT_syslog
  • syslog_develop_phase1
  • merge-asterisk-20.8.1
  • asterisk_rdkb_temp
  • 22.0.0-pre1
  • 21.4.2
  • 20.9.2
  • 18.24.2
  • certified-20.7-cert2
  • certified-18.9-cert11
  • 21.4.1
  • 20.9.1
  • 18.24.1
  • 21.4.0
  • 20.9.0
  • 18.24.0
  • certified-20.7-cert1
  • certified-18.9-cert10
  • 21.4.0-rc1
  • 20.9.0-rc1
  • 18.24.0-rc1
  • 21.3.1
  • 20.8.1
  • 18.23.1
40 results
Compare
user avatar
Add the ability to retrieve the source port of a SIP call.
Mark Michelson authored 
This adds the ability to call CHANNEL(recvport) on chan_sip
channels to see the port on which an INVITE was received.

ASTERISK-24040 #close
Reported by dtryba
Patches:
	dialplan_functions.patch uploaded by dtryba (License #6628)

Review: https://reviewboard.asterisk.org/r/3781



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ca725e1c
History
user avatar ca725e1c
History