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...---- * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write contents to file (Reported by Ray Crumrine) * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel Journo) * ASTERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) Bugs fixed in this release: ----------------------------------- * ASTERISK-25849 - chan_pjsip: transfers with direct media sometimes drops audio (Reported by Kevin Harwell) * ASTERISK-25113 - install_prereq in Debian 8 without "standard system utilities" (Reported by Rodri...

...---- * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write contents to file (Reported by Ray Crumrine) * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel Journo) * ASTERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) Bugs fixed in this release: ----------------------------------- * ASTERISK-25849 - chan_pjsip: transfers with direct media sometimes drops audio (Reported by Kevin Harwell) * ASTERISK-25113 - install_prereq in Debian 8 without "standard system utilities" (Reported by Rodri...

...asterisk/buildopts.h (Reported by Corey Farrell) * ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time. (Reported by Denis Alberto Martinez) * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls (Reported by Steve Davies) * ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by not here) * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax r...

...asterisk/buildopts.h (Reported by Corey Farrell) * ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time. (Reported by Denis Alberto Martinez) * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls (Reported by Steve Davies) * ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by not here) * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax r...

...UG are both enabled. (Reported by Corey Farrell) * ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time. (Reported by Denis Alberto Martinez) * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25037 - res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message (Reported by Joshua Colp) * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls (Reported by Steve Davies) * ASTERISK-22790 - check_modem_rate() may return...

...UG are both enabled. (Reported by Corey Farrell) * ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time. (Reported by Denis Alberto Martinez) * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25037 - res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message (Reported by Joshua Colp) * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls (Reported by Steve Davies) * ASTERISK-22790 - check_modem_rate() may return...

...bility to intercept hold and raise an event (Reported by Matt Jordan) * ASTERISK-25479 - Allow CDR's to be modified before being dispatched to engines (Reported by Jonh Wendell) * ASTERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID to something more palatable (Reported by Mark Michelson) * ASTERISK-25252 - ARI: Add the ability to manipulate log channels (Reported by Matt Jordan) * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (R...

...---- * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write contents to file (Reported by Ray Crumrine) * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel Journo) * ASTERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25419 - Dialplan Application for Integration of StatsD (Reported by Ashley Sanders) * ASTERISK-25549 - Confbridge: Add participant timeout option (Reported by Mark Michelson) * ASTERISK-24922 - ARI: Add the ability to intercept hold and raise an event (Reported by M...

...ncorrect use of strlen() leads to data corruption (Reported by Gianluca Merlo) * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local channel (Reported by Filip Frank) * ASTERISK-25800 - [patch] Calculate talktime when is first call answered (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes returns garbage (Reported by Etienne Lessard) * ASTERISK-20987 - non-admin users, who join muted conference are not being muted (Reported by hristo) * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST...

On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote: > ------------ > Requirements > ------------ > ....... > Speech API key from Google Yes... OK... but... where and how can I obtain this API Key? Where and how do I install it into my Asterisk box?

I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ncorrect use of strlen() leads to data corruption (Reported by Gianluca Merlo) * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local channel (Reported by Filip Frank) * ASTERISK-25800 - [patch] Calculate talktime when is first call answered (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes returns garbage (Reported by Etienne Lessard) * ASTERISK-20987 - non-admin users, who join muted conference are not being muted (Reported by hristo) * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST...

Hi all, I've got a new server up, but it's not staying up.... After a day or so, it segfaults with: [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.2(a) Driver]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

...create "dahdi_group" at CHANNEL function (Reported by Cirillo Ferreira) * ASTERISK-28279 - Added creation timestamp for bridge (Reported by sungtae kim) * ASTERISK-27483 - Allow wrapuptime to be set for each queue member (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-28055 - app_queue: Per-member wrapup time missing from AddQueueMember application (Reported by Niksa Baldun) * ASTERISK-28292 - Changed to show all channel stats including wrong media (Reported by sungtae kim) * ASTERISK-28253 - res_pjsip_session: Adding rtcp...

...- Segfault in app_queue on transfer scenarios (Reported by Etienne Lessard) * ASTERISK-25353 - [patch] Transcoding while different in Frame size = Frames lost (Reported by Alexander Traud) * ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25390 - default_from_user can crash with certain configuration backends (Reported by Mark Michelson) * ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten (Reported by Matt Jordan) * ASTERISK-25227 - No audi...

Dear list, i have a issue Asterisk crash (Module res_odbc exactly) after the same log who is "/ERROR[23805] astobj2.c: bad magic number.../" you will see on the log : Today [2016-02-24 16:00:38] ERROR[23805] *astobj2.c: bad magic number 0x552f302e for 0x7fe3505b3958* [2016-02-24 16:00:44] Asterisk 11.2-cert1 built by root @ Voice_server on a x86_64 running Linux on 2013-04-09

...a bug,but maybe the codeblock will never run (Reported by Huangyx) * ASTERISK-27277 - bridge: Renegotiate if source stream changes. (Reported by Joshua Colp) * ASTERISK-27283 - Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-27264 - res_pjsip_session: Crashes after sending PRACK and receiving 200 OK (Reported by Daniel Heckl) * ASTERISK-27260 - [pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36 (Reported by Daniel Heckl) * ASTERISK-27257 - bridge_native_rtp:...

...C2833 (SSRC Changes) (Reported by Ross Beer) * ASTERISK-27289 - A codeblock that maintains a bug,but maybe the codeblock will never run (Reported by Huangyx) * ASTERISK-27283 - Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-27257 - bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis - Prescom) * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability (Reported by Ross Beer) * ASTERISK-26606 - tcptls:...

...C2833 (SSRC Changes) (Reported by Ross Beer) * ASTERISK-27289 - A codeblock that maintains a bug,but maybe the codeblock will never run (Reported by Huangyx) * ASTERISK-27283 - Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-27257 - bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis - Prescom) * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability (Reported by Ross Beer) * ASTERISK-26606 - tcptls:...