September 2014 Archives by thread
      
      Starting: Mon Sep  1 09:07:17 CDT 2014
         Ending: Tue Sep 30 18:15:13 CDT 2014
         Messages: 514
     
- [asterisk-dev] [Code Review] 3929: ARI: Holding bridge issues with bridge Music on Hold, playback operations, and default channel roles
 
Matt Jordan
- [asterisk-dev] Need help to add ECT service based on ITU standard
 
babak
- [asterisk-dev] [Code Review] 3964: CDRs: Fix crash caused by infinite generation of new CDR records when a channel enters, leaves, and enters a multi-party bridge
 
Matt Jordan
- [asterisk-dev] [Code Review] 3965: testsuite: Test CDRs in a	multi-party bridge
 
Matt Jordan
- [asterisk-dev] [Code Review] 3863: testsuite: Add two tests for transmission of RTCP information to a HEP server
 
Matt Jordan
- [asterisk-dev] Issue AGI Get Full Variable
 
Bryant Zimmerman
- [asterisk-dev] [Code Review] 3966: Testsuite: RLS batched	notification tests
 
Mark Michelson
- [asterisk-dev] [Code Review] 3930: PJSIP: Resolve race condition regarding media handling when receiving 200 OK
 
Mark Michelson
- [asterisk-dev] [Code Review] 3967: Subscriptoin state test events	for review 3966
 
Mark Michelson
- [asterisk-dev] AstriDevCon Reminder! Tuesday, October 21st, 2014
 
Rusty Newton
- [asterisk-dev] [Code Review] 3968: Dial API: Add an option to indicate that a dial is being used to replace the dialing channel from a bridge
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3969: Manager: FullyBooted events are sent to AMI users that log in even if they don't have system level read permission.
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3970: res_phoneprov: Refactor phoneprov to allow pluggable config providers.
 
George Joseph
- [asterisk-dev] [Code Review] 3971: astobj2.c/refcounter.py: Fix to deal with invalid object refs.
 
rmudgett
- [asterisk-dev] Fwd: Identifying the multiple cards Digium TE820
 
George Karagheaur
- [asterisk-dev] [Code Review] 3972: Change DAHDI_UDEV_HOOK_DIR to	honor --prefix
 
David Lee
- [asterisk-dev] [Code Review] 3954: pjsip_options.c: Fix race condition stopping periodic out of dialog OPTIONS request.
 
rmudgett
- [asterisk-dev] [Code Review] 3962: CDRs: preserve context/extension when executing a Macro or GoSub
 
rmudgett
- [asterisk-dev] [Code Review] 3963: testsuite: Add a test that verifies CDRs with a Dial embedded in a subroutine/macro
 
rmudgett
- [asterisk-dev] [Code Review] 3951: Testsuite: Nominal test for Inter-Asterisk MWI SIP publication
 
opticron
- [asterisk-dev] [Code Review] 3957: Update commit_msg.py To Work With A Web Proxy, Follow New Commit Msg Format, Prompt For License
 
opticron
- [asterisk-dev] [Code Review] 3975: Menuselect: Fix incorrect	enabling on failed deps
 
opticron
- [asterisk-dev] [Code Review] 3961: Testsuite: Off-nominal resource list subscription tests (Lists only, no lists of lists)
 
opticron
- [asterisk-dev] [Code Review] 3976: New module: res_pjsip_phoneprov_provider provides the integration between res_pjsip and res_phoneprov.
 
George Joseph
- [asterisk-dev] AstriCon Hackathon
 
Matthew Jordan
- [asterisk-dev] [Code Review] 3977: RLS: Pre-allocate transmission data buffer to allow for sending of large NOTIFY requests.
 
Mark Michelson
- [asterisk-dev] [Code Review] 3978: Testsuite: Test for RLS large	NOTIFY requests
 
Mark Michelson
- [asterisk-dev] [Code Review] 3979: Call IDs: channel Call ID appears as gibberish when shown via CLI command core show channel for a channel that doesn't have call ID set
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3980: cel_odbc: Add microseconds precision in the eventtime column
 
Etienne Lessard
- [asterisk-dev] [Code Review] 3960: res_pjsip_pubsub: Check supported headers for eventlist before allowing subscribe to resource list
 
Matt Jordan
- [asterisk-dev] [Code Review] 3924: Testsuite: Add modular event	action framework
 
Scott Griepentrog
- [asterisk-dev] Defining a call under the new bridging architecture
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3982: res_rtp_asterisk: Fix a slew of	TURN issues.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3981: chan_rtp: Add unicast RTP	support to chan_multicast_rtp.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3983: func_channel: Add CHANNEL(onhold) item to get the current hold status of the channel.
 
rmudgett
- [asterisk-dev] [Code Review] 3984: Module to add devstate for MWI.
 
Jason Parker
- [asterisk-dev] ADAT Not Initializing
 
Syed Sujahid
- [asterisk-dev] [Code Review] 3985: realtime configuration: anything that goes through ast_destroy_realtime_fields crashes if only a single key/value pair is used.
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3986: config: bug: fix truncation of included config files on permissions error
 
George Joseph
- [asterisk-dev] [Code Review] 3987: Bridging: Fix bouncing native	bridge
 
opticron
- [asterisk-dev] SIPit 31
 
Matthew Jordan
- [asterisk-dev] [Code Review] 3988: res_pjsip: Don't require a password when doing userpass authentication
 
Sean Bright
- [asterisk-dev] [Code Review] 3989: utils: Create ast_strsep function that ignores separators inside quotes.
 
George Joseph
- [asterisk-dev] chan_sip.c:23647 handle_request_invite: Failed to	authenticate device
 
Deepak Bhatia
- [asterisk-dev] Timeline ZRTP Implementation
 
Jonathan Brown
- [asterisk-dev] [Code Review] 3990: CDRs/Dial: Fix an assertion caused by advancing a neutral state channel straight into dial pending without going through dial
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3991: musiconhold: Add sort=randstart,	and deprecate old stuff.
 
wdoekes
- [asterisk-dev] udptl ports
 
James Cloos
- [asterisk-dev] [Code Review] 2964: res_pjsip_outbound_registration: Add "virtual line" support for automatic inbound matching
 
Joshua Colp
- [asterisk-dev] [Code Review] 3992: res_pjsip_sdp_rtp: Add	optimistic SRTP support.
 
Joshua Colp
- [asterisk-dev] [Code Review] 3926: sip.conf: Clarify that sipdebug=yes cannot be undone by the CLI
 
wdoekes
- [asterisk-dev] [Code Review] 3993: config: bug: Fix SEGV in ast_category_insert when a matching category isn't found
 
George Joseph
- [asterisk-dev] [Code Review] 2764: Add edit-MASTER-channel option to CHANNEL() function. To work around channel optimization removing my channel settings.
 
wdoekes
- [asterisk-dev] [Code Review] 3994: Bridges User Documentation Update
 
opticron
- [asterisk-dev] [Code Review] 3916: Testsuite: Add test for	AllVariables Status parameter
 
Matt Jordan
- [asterisk-dev] [Code Review] 3995: res_pjsip_endpoint_identifier_ip: Can't parse identify with match value containing CIDR
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3996: Fix zero duration vm description when using MixMonitor m option
 
Scott Griepentrog
- [asterisk-dev] Realtime ODBC voicemail and fromstring variable
 
Leandro Dardini
- [asterisk-dev] Git Migration
 
Matthew Jordan
- [asterisk-dev] [Code Review] 3999: chan_iax2: Jitterbuffer causes crash in Asterisk 13 on account of format changes
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3998: res_pjsip: ami: Fix error in AMI output when no transport is associated to an endpoint
 
George Joseph
- [asterisk-dev] [Code Review] 4000: res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.
 
rmudgett
- [asterisk-dev] Asterisk 11.6-cert6, 11.12.1,	12.5.1 Now Available (Security Release)
 
Asterisk Development Team
- [asterisk-dev] AST-2014-009: Remote crash based on malformed SIP	subscription requests
 
Asterisk Security Team
- [asterisk-dev] AST-2014-010: Remote crash when handling out of call	message in certain dialplan configurations
 
Asterisk Security Team
- [asterisk-dev] [Code Review] 4001: PJSIP: Prevent T38 framehook	being put on wrong channel
 
opticron
- [asterisk-dev] [Code Review] 3673: RLS: Nominal list tests
 
Mark Michelson
- [asterisk-dev] [Code Review] 4003: 503 incorrectly sent to	rentransmitted invites
 
Torrey Searle
- [asterisk-dev] [Code Review] 4006: Test to validate 503 not generated on INVITE retransmissions
 
Torrey Searle
- [asterisk-dev] [Code Review] 4007: testsuite: pick test suite temp	dir based on free space
 
Scott Griepentrog
- [asterisk-dev] asterisk current call number caller
 
mjabi mjabi
- [asterisk-dev] [Code Review] 4008: res_pjsip_session: Add additional checks to prevent session refresh in unpossible states.
 
Joshua Colp
- [asterisk-dev] Asterisk 1.8.31.0-rc1 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 11.13.0-rc1 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 13.0.0-beta2 Now Available!
 
Asterisk Development Team
- [asterisk-dev] Asterisk 12.6.0-rc1 Now Available
 
Asterisk Development Team
- [asterisk-dev] Opinions Needed: PJSIP Outboud Registration with	multiple server_uris
 
George Joseph
- [asterisk-dev] [Code Review] 4010: General option to musiconhold.conf, to make applications MOH override channels musicclass.
 
Kristian Høgh
- [asterisk-dev] [Code Review] 4012: testsuite: Allow multiple -t	options to be passed.
 
wdoekes
- [asterisk-dev] ICE failure with Chrome, multi-homed Asterisk
 
Daniel Pocock
- [asterisk-dev] beta2 compile failure
 
Paul Albrecht
- [asterisk-dev] [Code Review] 4013: Alembic: Add 'outgoing' enum value to sippeers directmedia enumerator
 
Jonathan Rose
- [asterisk-dev] [Code Review] 4014: Changes to CDR and CEL unit	tests to prevent FRACKs.
 
Mark Michelson
- [asterisk-dev] [Code Review] 4015: Get rid of most old libc free/malloc/realloc and replace with ast_free and friends.
 
wdoekes
- [asterisk-dev] DAHDI-Linux and DAHDI-Tools 2.10.0.1 Now Available
 
Asterisk Development Team
- [asterisk-dev] [Code Review] 4016: chan_sip: Unref outbound proxy structure on dialog(pvt) struct
 
wdoekes
- [asterisk-dev] [Code Review] 3948: Asterisk does not respect outbound proxy when sending qualify requests
 
wdoekes
- [asterisk-dev] [Code Review] 4017: chan_pjsip: Don't attempt to apply formats if there aren't any capabilities defined when creating a new channel
 
Jonathan Rose
- [asterisk-dev] [Code Review] 3716: Weak Reference Containers
 
George Joseph
- [asterisk-dev] Opinions Needed: Case sensitivity in config file	section names
 
George Joseph
- [asterisk-dev] [Code Review] 4018: res_pjsip: Make transport cipher option accept a comma separated list of cipher names.
 
rmudgett
- [asterisk-dev] [Code Review] 4019: PJSIP: Handle defaults properly
 
opticron
- [asterisk-dev] Asterisk 1.8.31.0 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 11.13.0 Now Available
 
Asterisk Development Team
- [asterisk-dev] Asterisk 12.6.0 Now Available
 
Asterisk Development Team
- [asterisk-dev] [Code Review] 4020: RLS Tests: off nominal tests for lists of lists (MWI and presence)
 
Jonathan Rose
- [asterisk-dev] [Code Review] 4023: Allow passing options and command to MixMonitor when recording in ConfBridge
 
gareth
- [asterisk-dev] [Code Review] 4025: pjsip_cli: Suppress header print	on error or no objects
 
George Joseph
- [asterisk-dev] [Code Review] 4024: AMI connection closes if sendMessage generates an erros response
 
Peter Katzmann
- [asterisk-dev] [Code Review] 4026: res_hep leaks reference to	configuration
 
Corey Farrell
- [asterisk-dev] unsubscribe
 
Curtis Raams
- [asterisk-dev] [Code Review] 4031: Realtime peers are never unref:	memory leaks
 
ibercom
- [asterisk-dev] [Code Review] 4032: PJSIP: Force transport on	contact rewrite
 
opticron
- [asterisk-dev] Confbridge Performance using G722 @ max 3 Party in 1	conference.
 
bala murugan
- [asterisk-dev] [Code Review] 4033: manager/config: Enhancements to support templates and non-unique category names via AMI
 
George Joseph
- [asterisk-dev] [Code Review] 4034: chan_pjsip: Fix deadlock when masquerading PJSIP channels.
 
rmudgett
- [asterisk-dev] [Code Review] 4035: Dialplan function to get first/head caller channel on queue
 
Kristian Høgh
- [asterisk-dev] [Code Review] 4036: pjsip sample: suggest use of rewrite_contact for natted endpoints
 
Scott Griepentrog
    
      Last message date: 
       Tue Sep 30 18:15:13 CDT 2014
    Archived on: Tue Sep 30 18:14:18 CDT 2014
    
   
     
     
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