Re: [asterisk-users] 401 error
On Thu, 9 Mar 2023, Jerry Geis wrote: Trying to setup an incoming call with a DNIS When I dial the number - I see nothing on the CLI. Have you enabled [PJ]SIP debugging? Bumping up console debug and verbose levels may also yield clues. tcpdump+sngrep are my 'gotos' for packet analysis, but this may not need too much depth. The person says the server is returning 401 How do I debug that. Using asterisk 18.8.0 https://en.wikipedia.org/wiki/List_of_SIP_response_codes#:~:text=401%20Unauthorized,1%5D%3A%E2%80%8A%C2%A721.4.2 "401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars.[1]: §21.4.2" My guess would be a user or password mismatch. Are you using SIP or PJSIP? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 401 error
I have a SIP trunk - calls going out work fine. Trying to setup an incoming call with a DNIS When I dial the number - I see nothing on the CLI. The person says the server is returning 401 How do I debug that. Using asterisk 18.8.0 Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 20.2.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 20.2.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 20.2.0 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: New Features made in this release: --- * ASTERISK-29810 - app_signal: Add channel signaling applications (Reported by N A) * ASTERISK-30262 - res_pjsip_session: Allow a context to be specified for overlap dialing (Reported by N A) * ASTERISK-30319 - Add BYE Reason support for SIP (Reported by Igor Goncharovsky) * ASTERISK-30180 - app_broadcast: Add a channel audio multicasting application (Reported by N A) Bugs fixed in this release: --- * ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string (Reported by AvayaXAsterisk) * ASTERISK-30354 - chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall (Reported by N A) * ASTERISK-30162 - when chan_iax is used to relay calls, no ringing indication is played (Reported by Jaco Kroon) * ASTERISK-30424 - pjproject_bundled: cross-compilation broken when ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-29604 - ari: Segfault with lots of calls (Reported by Danila Evgrafov) * ASTERISK-30406 - pbx_ael: Global variables are not expanded. (Reported by Sean Bright) * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding g722 after MES changes (Reported by George Joseph) * ASTERISK-30345 - loader.c: Modules that decline to load cannot be reloaded (Reported by N A) * ASTERISK-30351 - manager: Originate variables are not added when setvar used in manager.conf (Reported by Sebastian Gutierrez) * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down when they shouldn't be (Reported by Joshua C. Colp) * ASTERISK-30379 - http: fix NULL pointer dereference while enable_status on TLS-only (Reported by Boris P. Korzun) * ASTERISK-30375 - res_http_media_cache: Crash when URL has no path component. (Reported by Sean Bright) * ASTERISK-30367 - pbx: Fix outdated channel snapshots with pbx_exec (Reported by N A) * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late (Reported by Oleg) * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold (Reported by Benjamin Keith Ford) * ASTERISK-30240 - app voicemail odbc build error with gcc 11.1 (Reported by Michael Bradeen) * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to endpoint (Reported by Yury Kirsanov) * ASTERISK-30198 - Error `Too many open files` occurs after about ~8000 calls when using mixmonitor (Reported by Julien Alie) * ASTERISK-30347 - xmldocs: Remove references to removed applications (Reported by N A) Improvements made in this release: --- * ASTERISK-30411 - app_read: add option to include terminating digit on empty, terminated strings (Reported by Michael Bradeen) * ASTERISK-30405 - app_directory: Add 's' option to skip channel call (Reported by Michael Bradeen) * ASTERISK-30422 - app_senddtmf: add the option for senddtmf to answer (Reported by Michael Bradeen) * ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13 (Reported by Stanislav Abramenkov) * ASTERISK-30404 - app_directory: Add reading directory configuration from custom file (Reported by Michael Bradeen) * ASTERISK-29913 - func_json: Adds multi-level and array parsing to JSON_DECODE (Reported by N A) * ASTERISK-30353 - func_frame_trace: Print text for text frames (Reported by N A) * ASTERISK-30361 - json.h: Add missing ast_json_object_real_get (Reported by N A) * ASTERISK-30280 - Create capability to assign a Media Experience Score to RTP streams (Reported by George Joseph) * ASTERISK-30332 - func_callerid: Warn if invalid redirecting reason provided (Reported by N A) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0 Thank you for your continued support of
[asterisk-users] Asterisk 18.17.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 18.17.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.17.0 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: New Features made in this release: --- * ASTERISK-29810 - app_signal: Add channel signaling applications (Reported by N A) * ASTERISK-30262 - res_pjsip_session: Allow a context to be specified for overlap dialing (Reported by N A) * ASTERISK-30319 - Add BYE Reason support for SIP (Reported by Igor Goncharovsky) * ASTERISK-30180 - app_broadcast: Add a channel audio multicasting application (Reported by N A) Bugs fixed in this release: --- * ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string (Reported by AvayaXAsterisk) * ASTERISK-30354 - chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall (Reported by N A) * ASTERISK-30162 - when chan_iax is used to relay calls, no ringing indication is played (Reported by Jaco Kroon) * ASTERISK-30424 - pjproject_bundled: cross-compilation broken when ssl autodetected (Reported by Nick French) * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when multi-homed (Reported by cmaj) * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by Ross Beer) * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember (Reported by Sean Bright) * ASTERISK-30406 - pbx_ael: Global variables are not expanded. (Reported by Sean Bright) * ASTERISK-29604 - ari: Segfault with lots of calls (Reported by Danila Evgrafov) * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding g722 after MES changes (Reported by George Joseph) * ASTERISK-30345 - loader.c: Modules that decline to load cannot be reloaded (Reported by N A) * ASTERISK-30379 - http: fix NULL pointer dereference while enable_status on TLS-only (Reported by Boris P. Korzun) * ASTERISK-30375 - res_http_media_cache: Crash when URL has no path component. (Reported by Sean Bright) * ASTERISK-30351 - manager: Originate variables are not added when setvar used in manager.conf (Reported by Sebastian Gutierrez) * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down when they shouldn't be (Reported by Joshua C. Colp) * ASTERISK-30367 - pbx: Fix outdated channel snapshots with pbx_exec (Reported by N A) * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late (Reported by Oleg) * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold (Reported by Benjamin Keith Ford) * ASTERISK-30240 - app voicemail odbc build error with gcc 11.1 (Reported by Michael Bradeen) * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to endpoint (Reported by Yury Kirsanov) * ASTERISK-30198 - Error `Too many open files` occurs after about ~8000 calls when using mixmonitor (Reported by Julien Alie) Improvements made in this release: --- * ASTERISK-30411 - app_read: add option to include terminating digit on empty, terminated strings (Reported by Michael Bradeen) * ASTERISK-30405 - app_directory: Add 's' option to skip channel call (Reported by Michael Bradeen) * ASTERISK-30422 - app_senddtmf: add the option for senddtmf to answer (Reported by Michael Bradeen) * ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13 (Reported by Stanislav Abramenkov) * ASTERISK-30404 - app_directory: Add reading directory configuration from custom file (Reported by Michael Bradeen) * ASTERISK-29913 - func_json: Adds multi-level and array parsing to JSON_DECODE (Reported by N A) * ASTERISK-30353 - func_frame_trace: Print text for text frames (Reported by N A) * ASTERISK-30361 - json.h: Add missing ast_json_object_real_get (Reported by N A) * ASTERISK-30280 - Create capability to assign a Media Experience Score to RTP streams (Reported by George Joseph) * ASTERISK-30332 - func_callerid: Warn if invalid redirecting reason provided (Reported by N A) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth
