[asterisk-users] PJSIP wizard reload not reloading ?
Hi list, I'm having a strange problem when using pjsip wizard and reloading ("pjsip reload" on CLI): some data (specifically endpoint/pickup_group) is not modified. For example, initially I have empty pickup group: tiare*CLI> pjsip show endpoint xxx ... pickup_group : ... Then, I add endpoint/pickup_group = 0,3 to pjsip_wizard.conf, and reload: pickup_group remains empty. Then, if I change the line in pjsip_wizard.conf to endpoint/pickup_group = 0, 3 ^ note the space here! then reload, and I get what was expected: tiare*CLI> pjsip show endpoint xxx ... pickup_group : 0, 3 ... I have seen this problem on Asterisk-16 only (up to latest 16.5.0). The modified configuration file is included from /etc/asterisk/pjsip_wizard.conf: #include astportal/pjsip_wizard.conf pjsip reload has default definition in cli_aliases.conf: pjsip reload=module reload res_pjsip.so res_pjsip_authenticator_digest.so res_pjsip_endpoint_identifier_ip.so res_pjsip_mwi.so res_pjsip_notify.so res_pjsip_outbound_publish.so res_pjsip_publish_asterisk.so res_pjsip_outbound_registration.so Did I miss something, or should I open an issue? Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 signature.asc Description: OpenPGP digital signature -- _ -- 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] Male French Talent
Hi, Does anyone out there know of male french talent for Asterisk sound files where the talent already recorded the bulk of the Asterisk sound files? TIA. Regards, Dovid -- _ -- 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
Re: [asterisk-users] Wanted: professional softphone
look at zoiper oem.zoiper.com you could create a url that creates a build with all credentials -- _ -- 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] Calls and queue statistics
Hello list, I'm looking for a solution that can be applied to a stock asterisk 16 (pjsip if it matter) running Debian 9 (php7.0). Statistics should be available for normal calls and queues using a WEB interface. Open source better but not necessary, Any feedback appreciate, no matter if it's a "go for it" or "go away". Regards -- Daniel -- _ -- 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
Re: [asterisk-users] Wanted: professional softphone
On 7/24/19 11:41 PM, Michael Maier wrote: Hello! Does anybody by chance know of a softphone which additionally has a management suite to fully configure it userID based for Windows clients? Any idea is appreciated! Zulu from Sangoma allows you to generate a QR code that configures everything automatically for each user. Been using it lately and it works very well. Only downside is that it is only for FreePBX/PBXact. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- 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 16.5.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 16.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.5.0 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: Security bugs fixed in this release: --- * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with no body causes crash (Reported by Gil Richard) * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38 reINVITE (Reported by Francesco Castellano) Bugs fixed in this release: --- * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317 (Reported by abelbeck) * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable (Reported by Michael Maier) * ASTERISK-26006 - Show offending IP for TLS setup failures in logs (Reported by Oleksandr Natalenko) * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged (Reported by Bernhard Schmidt) * ASTERISK-28419 - app_amd: Does not work with silence suppression (Reported by Nasir Iqbal) * ASTERISK-28018 - IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate (Reported by vijay kumar) * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event (Reported by Abhay Gupta) * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry Svyatogorov) * ASTERISK-27981 - res_fax: Fax session leak with fax gatewaying (Reported by pasandev) * ASTERISK-28427 - new mwi.h include missing from some dahdi source files, causes build failure (Reported by Guido Falsi) * ASTERISK-28421 - Wrong type used for timestamp in res_rtp_asterisk (Reported by Morten Tryfoss) * ASTERISK-27994 - PJSIP: Early media ringback not indicated after Progress() (Reported by Gregory Massel) Improvements made in this release: --- * ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi (Reported by Kirsty Tyerman) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0 Thank you for your continued support of Asterisk! -- _ -- 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 13.28.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 13.28.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.28.0 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: Security bugs fixed in this release: --- * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with no body causes crash (Reported by Gil Richard) * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38 reINVITE (Reported by Francesco Castellano) Bugs fixed in this release: --- * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317 (Reported by abelbeck) * ASTERISK-26006 - Show offending IP for TLS setup failures in logs (Reported by Oleksandr Natalenko) * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged (Reported by Bernhard Schmidt) * ASTERISK-28460 - res_pjsip_sdp_rtp: Fix ICE candidate leak with specific usage (Reported by Joshua C. Colp) * ASTERISK-28018 - IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate (Reported by vijay kumar) * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event (Reported by Abhay Gupta) * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry Svyatogorov) * ASTERISK-27981 - res_fax: Fax session leak with fax gatewaying (Reported by pasandev) * ASTERISK-28419 - app_amd: Does not work with silence suppression (Reported by Nasir Iqbal) * ASTERISK-28427 - new mwi.h include missing from some dahdi source files, causes build failure (Reported by Guido Falsi) * ASTERISK-27994 - PJSIP: Early media ringback not indicated after Progress() (Reported by Gregory Massel) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.28.0 Thank you for your continued support of Asterisk! -- _ -- 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