search for: katsnelson

Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunking. Cateris paribus, what is the preferred solution to choose? What points to consider? I can name the provider if this is not against this list policy--is it? Thanks, -kkm

...endpoint (Reported by Ross Beer) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-2...

...endpoint (Reported by Ross Beer) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-2...

I have a private trunk with a peer. A call comes in from the trunk, and Asterisk calls a peer agent. If the agent transfers the call (this is a blind transfer using REFER), I want Asterisk to send a REFER back to the trunk, and essentially stay out of the loop. As set up, Asterisk initiates a new call using INVITE to the trunk and then bridges the original incoming and the new outgoing calls.

Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped? -kkm

...an't contain commas (Reported by S��bastien Duthil) * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them (Reported by test011) * ASTERISK-17799 - AEL reload causes loss of control in a macro (Reported by Kirill Katsnelson) * ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter (Reported by Luke-Jr) * ASTERISK-14939 - AEL parsers does not find existing label (Reported by klaus3000) * ASTERISK-20182 - Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIf...

...an't contain commas (Reported by S��bastien Duthil) * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them (Reported by test011) * ASTERISK-17799 - AEL reload causes loss of control in a macro (Reported by Kirill Katsnelson) * ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter (Reported by Luke-Jr) * ASTERISK-14939 - AEL parsers does not find existing label (Reported by klaus3000) * ASTERISK-20182 - Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIf...

https://bugzilla.mindrot.org/show_bug.cgi?id=2246 Bug ID: 2246 Summary: PAM enhancements for OpenSSH server Product: Portable OpenSSH Version: 6.6p1 Hardware: Sparc OS: Solaris Status: NEW Severity: enhancement Priority: P5 Component: PAM support Assignee: unassigned-bugs at

...ported by nik600) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) New Features made in this release: ----------------------------------- * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by Alexei Gradinari) For...

...ported by nik600) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) New Features made in this release: ----------------------------------- * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by Alexei Gradinari) For...

...ash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson)...

...an't contain commas (Reported by S��bastien Duthil) * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them (Reported by test011) * ASTERISK-17799 - AEL reload causes loss of control in a macro (Reported by Kirill Katsnelson) * ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter (Reported by Luke-Jr) * ASTERISK-14939 - AEL parsers does not find existing label (Reported by klaus3000) * ASTERISK-20182 - Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIf...

...an't contain commas (Reported by S��bastien Duthil) * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them (Reported by test011) * ASTERISK-17799 - AEL reload causes loss of control in a macro (Reported by Kirill Katsnelson) * ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter (Reported by Luke-Jr) * ASTERISK-14939 - AEL parsers does not find existing label (Reported by klaus3000) * ASTERISK-20182 - Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIf...

...ion (Reported by Thomas Sevestre) * ASTERISK-27322 - [New Feature] Add mute and DTMF passthrough to ARI add channel to bridge (Reported by Darren Sessions) * ASTERISK-27162 - [patch]chan_sip: Access incoming SIP REFER headers in the dialplan (Reported by Kirill Katsnelson) * ASTERISK-27163 - chan_sip: Dialplan function SIP_HEADERS() to complement SIP_HEADER(). (Reported by Kirill Katsnelson) * ASTERISK-27063 - Add support for systemd socket activation (Reported by Corey Farrell) Bugs fixed in this release: -----------------------------------...

...ash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier) *...

...ash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier) *...

...an't contain commas (Reported by S��bastien Duthil) * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them (Reported by test011) * ASTERISK-17799 - AEL reload causes loss of control in a macro (Reported by Kirill Katsnelson) * ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter (Reported by Luke-Jr) * ASTERISK-14939 - AEL parsers does not find existing label (Reported by klaus3000) * ASTERISK-20182 - Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIf...

...ion (Reported by Thomas Sevestre) * ASTERISK-27322 - [New Feature] Add mute and DTMF passthrough to ARI add channel to bridge (Reported by Darren Sessions) * ASTERISK-27162 - [patch]chan_sip: Access incoming SIP REFER headers in the dialplan (Reported by Kirill Katsnelson) * ASTERISK-27163 - chan_sip: Dialplan function SIP_HEADERS() to complement SIP_HEADER(). (Reported by Kirill Katsnelson) * ASTERISK-27063 - Add support for systemd socket activation (Reported by Corey Farrell) Bugs fixed in this release: -----------------------------------...

...ported by nik600) * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself (Reported by Jonathan R. Rose) * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus events for autocreated peers (Reported by Kirill Katsnelson) * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName (Reported by Javier Acosta) * ASTERISK-25927 - Removed option "registertrying" is still documented in sip.conf.sample (Reported by Etienne Lessard) * ASTERISK-25948 - ast_pthrea...

...ISK-27322>] - [New Feature] Add mute and DTMF passthrough to ARI add channel to bridge (Reported by Darren Sessions) - [ASTERISK-27162 <https://issues.asterisk.org/jira/browse/ASTERISK-27162>] - [patch]chan_sip: Access incoming SIP REFER headers in the dialplan (Reported by Kirill Katsnelson) - [ASTERISK-27163 <https://issues.asterisk.org/jira/browse/ASTERISK-27163>] - chan_sip: Dialplan function SIP_HEADERS() to complement SIP_HEADER(). (Reported by Kirill Katsnelson) - [ASTERISK-27129 <https://issues.asterisk.org/jira/browse/ASTERISK-27129>] - ast_waitford...