search for: res_pjsip_caller_id

...res_pjsip (see below) on the asterisk CLI. Anyone have some input on this ? Thanks. Kind regards. [root at sip admin]# ls /usr/lib64/asterisk/modules | grep pjsip chan_pjsip.so func_pjsip_aor.so func_pjsip_contact.so func_pjsip_endpoint.so res_pjsip_acl.so res_pjsip_authenticator_digest.so res_pjsip_caller_id.so res_pjsip_config_wizard.so res_pjsip_dialog_info_body_generator.so res_pjsip_diversion.so res_pjsip_dlg_options.so res_pjsip_dtmf_info.so res_pjsip_endpoint_identifier_anonymous.so res_pjsip_endpoint_identifier_ip.so res_pjsip_endpoint_identifier_user.so res_pjsip_exten_state.so res_pjsip_header...

...RISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information (Reported by George Joseph) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.1-cert7 Thank you for your continued support of Aster...

...tifier_ip.so and it is loaded. In fact, all others authentitcation modules are not, since I only identify via IP. I did not load all the methods I will never use. load => res_pjsip.so load => chan_pjsip.so load => res_pjsip_acl.so noload => res_pjsip_authenticator_digest.so load => res_pjsip_caller_id.so load => res_pjsip_diversion.so load => res_pjsip_dtmf_info.so noload => res_pjsip_endpoint_identifier_anonymous.so load => res_pjsip_endpoint_identifier_ip.so noload => res_pjsip_endpoint_identifier_user.so load => res_pjsip_exten_state.so load => res_pjsip_log_forwarder.so...

...RISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information (Reported by George Joseph) * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP thread (Reported by Joshua Colp) * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised (Reported by Joshua Co...

...this release: ----------------------------------- * ASTERISK-25495 - [patch] Prevent old-update packages on repository Debian systems (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps (Reported by Andrew Nagy) * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for Anonymous <anonymous at anonymous.invalid> (Reported by Anthony Messina) * ASTERISK-24813 - asterisk.c: #if statement in listener() confuses code folding editors (Reported by Corey Farrell) * ASTERISK-25767 - [patch] Add check to configure for sanitize...

...K-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel)...

...K-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel)...

...this release: ----------------------------------- * ASTERISK-25495 - [patch] Prevent old-update packages on repository Debian systems (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps (Reported by Andrew Nagy) * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for Anonymous <anonymous at anonymous.invalid> (Reported by Anthony Messina) * ASTERISK-24813 - asterisk.c: #if statement in listener() confuses code folding editors (Reported by Corey Farrell) * ASTERISK-25767 - [patch] Add check to configure for sanitize...

...K-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel)...

...RISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information (Reported by George Joseph) * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP thread (Reported by Joshua Colp) * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised (Reported by Joshua Co...

...RISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information (Reported by George Joseph) * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP thread (Reported by Joshua Colp) * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised (Reported by Joshua Co...

...d by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported b...

...d by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported b...

...(Reported by Matt Jordan) - [ASTERISK-26480 <https://issues.asterisk.org/jira/browse/ASTERISK-26480>] - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) - [ASTERISK-26307 <https://issues.asterisk.org/jira/browse/ASTERISK-26307>] - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) - [ASTERISK-26503 <https://issues.asterisk.org/jira/browse/ASTERISK-26503>] - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) - [ASTERISK-26423 <https://issues.asterisk.org/jira/browse/A...