1523 Commits (0fef06f8ab99360521acdba3ec66aaa3558e4dbe)

Author	SHA1	Message	Date
Walter Doekes	2305602863	chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI. ... Document it in sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod Review: https://reviewboard.asterisk.org/r/3926/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@423066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	863e7e528c	chan_dahdi: Add inband_on_setup_ack compatibility option. ... The new inband_on_setup_ack option causes Asterisk to assume inband audio may be present when a SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a dialtone is sent from the network side, progress indicator 8 "Inband info now available" MAY be sent to the CPE if no digits were received with the SETUP. It is thus implied that the ie is mandatory if digits came with the SETUP and dialtone is needed. This option should be enabled, when the network sends dialtone and you want to hear it, but the network doesn't send the progress indicator when needed. NOTE: For Q.SIG setups this option should be enabled when outgoing overlap dialing is also enabled because Q.SIG does not send the progress indicator with the SETUP ACK. The commit -r413714 (AST-1338) which causes this issue was dealing with a SIP-to-ISDN interoperability issue. This commit is a merge of the two patches indicated below. ASTERISK-23897 #close Reported by: Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3633/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	d3d407c175	main/features - documentation - reformat examples and options in features.conf.sample to show clearly which options apply in which section ... The features.conf sample can be a bit confusing about what parking options can be set only in the general context, or both in the general context (for the default parking lot) and in other parking lot contexts. A bug was filed due to confusion and a little googling will show lots of other confused users. Despite some comments on the individual options, it still reads in a confusing way. In this patch I separate out those options with some headings in to attempt a better layout. I went ahead and modified other headings in the file, or added them to facilitate better visual scanning. ASTERISK-23667 #close Review: https://reviewboard.asterisk.org/r/3621/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	c2c3e17138	Update extensions.lua.sample with naming conflict guidance. ... The sample extensions.lua was causing pbx_lua to fail to load when parsing 'app.goto("default", "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This patch adds guidance to extensions.lua.sample and changed 'app.goto("default", "s", 1)' to 'app.['goto']("default", "s", 1)'. https://reviewboard.asterisk.org/r/3627/ ASTERISK-23844 #comment This commit fixes 1.8, patch for 11->trunk coming. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	7550fe7efb	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	5d924bc0fc	chan_sip: Add sendrpid trust options ... In r411189, some behavior was changed which made sendrpid behavior act in a more trusting manner by sending full user data for peers set with private caller presence in P-Asserted-Identity headers. Since this changed long time expected behaviors, we decided to pull that patch when that was pointed out by the community. Instead, this patch provides a trust_id_outbound setting which will expose the data per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers at all if set to 'no'. By default trust_id_outbound will be set to 'legacy' which will preserve the behavior prior to these patches. Extra special thanks to Walter Doekes for providing advice and feedback. (closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski Review: https://reviewboard.asterisk.org/r/3447/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	445b1ced10	Reverting r411189 so that it can be put up for public review ... --- r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior to this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) --- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	7dc87fa2b9	configs: Clean up long line and typo in res_odbc.conf.sample. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a997ddc829	internal_timing: Remove the option and always make it enabled if a timing module is loaded. ... The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f222e30437	res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values. ... This patch fixes setting nullable integer columns to NULL instead of an empty string, which fails for PostgreSQL, for example. The current code is supposed to do so, but the check is broken. The patch also allows the first column in the list to be a nullable integer. This patch also adds a compatibility setting in res_odbc.conf, allow_empty_string_in_nontext. It is enabled by default. It should be disabled for database backends (such as PostgreSQL) that require NULL instead of an empty string for Integer columns. Review: https://reviewboard.asterisk.org/r/3375 (issue ASTERISK-23459) Reported by: zvision patches: res_config_odbc.diff uploaded by zvision (License 5755) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	e2708b03b6	chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) ... Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	a6a92ffe3d	res_fax: Warn that minrate=2400 is not valid for V.27 instead of failing load. ... Change minrate from 2400 to 4800 on config reload in response to changes from ASTERISK-22790 only. Any config with minrate of 2400 that would fail before r405693 will still fail. Comment out many settings in res_fax.conf.sample. The defaults are set in res_fax.c, so setting the same value in sample config does nothing but make the sample config more fragile. (closes issue ASTERISK-23231) Reported by: David Brillert Review: https://reviewboard.asterisk.org/r/3261/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	d72f70e9a5	configs/voicemail.conf.sample - Make mailcmd sample text more explicit ... Made the wording a bit more explicit. Didn't really change the meaning. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	f0f80e418b	configs/agents.conf.sample - Remove example for non-functional "goodbye" parameter ... The "goodbye" parameter is not implemented in the source code, it does nothing. (closes issue SWP-6518) Reported By: Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6e6a7f2af0	chan_iax2: Block unnecessary control frames to/from the wire. ... Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later) results in an unexpected call disconnect. The problem happens because newer values in the enum ast_control_frame_type are not consistent between the branch versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) using IAX2 2) v1.8 answers and sends a connected line update control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 receives the control frame as an end-of-q (on v1.4 AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the receive queue becomes empty. Several things are done by this patch to fix the problem and attempt to prevent it from happening again in the future: * Added a warning at the definition of enum ast_control_frame_type about how to add new control frame values. * Made block sending and receiving control frames that have no reason to go over the wire. * Extended the connectedline iax.conf parameter to also include the redirecting information updates. * Updated the connectedline iax.conf parameter documentation to include a notice that the parameter must be "no" when the peer is an Asterisk v1.4 instance. (closes issue AST-1302) Review: https://reviewboard.asterisk.org/r/3174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	8878f38742	indications.conf: add stutter tone; end properly ... * If the "stutter" (voicemail indication) tone is indeed a stutter tone, and it ends with a constant tone, make sure that it is the dial tone. This was done for India (in), Mexico (mx) and the Philippines (ph). * If no "stutter" tone exists for a country, provide one. This was done for Spain (es), Malaysia (my) and Venezuela (ve). Review: https://reviewboard.asterisk.org/r/3158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d06ec403aa	tcptls.c: Made TLS handle a certificate chain file. ... Thanks to Guillaume Martres for doing the necessary research to validate the change. (closes issue ASTERISK-17727) Reported by: LN Patches: use_certificate_chain.patch (license #5864) patch uploaded by st documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	f82e6e93c5	queues.conf.sample Fix documented default for persistentmembers ... Closes issue ASTERISK-22662 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	b4023b47d1	manager: Clarify eventfilter documentation. Textual changes only. ... Review: https://reviewboard.asterisk.org/r/3133/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	66cce0927e	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 ... Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	affe507d5b	res_fax: check_modem_rate() returned incorrect rate for V.27 ... According to the new standard for V.27 and V.32 they are able to transmit at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be updated to reflect this. Also, because of this change the default 'minrate' value was updated to be 4800. (closes issue ASTERISK-22790) Reported by: Paolo Compagnini Patches: res_fax.txt uploaded by looserouting (license 6548) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	99267c7a7b	security: Inhibit execution of privilege escalating functions ... This patch allows individual dialplan functions to be marked as 'dangerous', to inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. (closes issue ASTERISK-22905) Review: http://reviewboard.digium.internal/r/432/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@403913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	6c77f74463	chan_sip: Allow Asterisk to retry after 403 on register ... This adds a global option in chan_sip to allow it to continue attempting registration if a 403 is received, clearing the cached nonce and treating it as a non-fatal response. Normally, this would cause registration attempts to that endpoint to stop. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	14344fb803	Documentation fix - waitfordialtone is not boolean, it's time in milliseconds ... Changing text in chan_dahdi.conf sample to be accurate. (issue ASTERISK-22308) (closes issue ASTERISK-22308) Reported By: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@398880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	803adef97e	Add "autoframing" option to sip.conf.sample and h323.conf.sample. ... The autoframing option was added to chan_sip.c in r43243 (mogorman, 2006-09-19 01:32:57), but never made its way into the sample configs. Review: https://reviewboard.asterisk.org/r/2768/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@396994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d1eee5436a	Provide proper ring tone in indications.conf for Malaysia ... The ring tone provided in the sample indications.conf was incorrect. This patch modifies the sample ring tone to be what it should: ring = 425/400,0/200,425/400,0/2000 This brings it in line with the tone definition in DAHDI 2.7.0. (zonedata.c) (closes issue ASTERISK-21997) Reported by: Filip Jenicek patches: malaysia_ring.patch uploaded by phill (License 6277) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@394940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	17fe92079e	Document connectedline parameter for chan_iax2 ... The connectedline parameter for a chan_iax2 peer was undocumented. This patch documents the options in the sample configuration file. (closes issue ASTERISK-21953) Reported by: Birger "WIMPy" Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@394886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	4a6c3a4237	astobj2-ify the SLA code ... The SLA code within app_meetme was written before asotbj2 had been merged into Asterisk. Worse, support for reloads did not exist at first and was added later as a bolt-on feature. I knew at the time that reloading was not safe at all while SLA was in use, so the reload would be queued up to execute when the system was idle. Unfortunately, this approach was still prone to errors beyond the fact that this was the only place in Asterisk where configuration was not reloaded instantly when requested. This patch converts various SLA objects to be reference counted objects using astobj2. This allows reloads to be processed while the system is in use. The code ensures that the objects will not disappear while one of the other threads is using them. However, they will be immediately removed from the global trunk and station containers so no new calls will use them if removed from configuration. Review: https://reviewboard.asterisk.org/r/2581/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@393928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0897f95b39	Add announce-to-first-user option for app_queue ... In r386792, the ability to play prompts to the first caller in a call queue was added. While this is arguably a bug fix for those who expect the first caller to continue receiving prompts while the agent is dialed, it has the side effect of preventing the first caller from hearing the agent immediately upon bridging. This may not be a problem for those who really want this option, but for those who didn't care whether or not the first caller in queue heard their position, it was an issue. This patch disables the ability for the first caller in the queue to hear prompts and adds a new option, announce-to-first-user, to queues.conf. Those who the behavior can enable it by setting this value to True. Note that if we ever implement the ability to have the prompts be stopped upon bridging, this option can be removed. (closes issue ASTERISK-21782) Reported by: Remi Quezada git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@391215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	ef5a22c5b0	Modified the list of keys for the driver backends for sake of sample clarity ... Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7bdeb23dd2	chan_dahdi: Add inband_on_proceeding compatibility option. ... The new inband_on_proceeding option causes Asterisk to assume inband audio may be present when a PROCEEDING message is received. Q.931 Section 5.1.2 says the network cannot assume that the CPE side has attached to the B channel at this time without explicitly sending the progress indicator ie informing the CPE side to attach to the B channel for audio. However, some non-compliant ISDN switches send a PROCEEDING without the progress indicator ie indicating inband audio is available and assume that the CPE device has connected the media path for listening to ringback and other messages. ASTERISK-17834 which causes this issue was dealing with a non-compliant network switch. (closes issue ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	706235d7ee	Remove "registertrying" and add "rtp_engine" from/to sip.conf.sample ... The "registertrying" option was removed in r343220. The "rtp_engine" option was added in r186078 but erroneously named "engine" in the sample. Note that there is no global sip setting for a different engine. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	559695d25a	dahdi.conf.sample: Add description for "buffers" setting. ... This contains an edited version of the patch originally created by John Bigelow. (closes issue ASTERISK-14435) Reported by: John Bigelow Patches: buffers.patch (license #5091) patch uploaded by John Bigelow 0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch (license #5417) patch uploaded by Shaun Ruffell Modified git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	5b6aa46f43	dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END ... Instead of a recompile, allow values to be adjusted in dsp.conf For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2144/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	57e403e023	dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values ... Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries. Various countries have different specifications for the maximum power level differences between the DTMF low group and high group of frequencies. Power level difference between frequencies for different Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03) Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T specifications Add's the following variables to dsp.conf ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51 ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98 (closes issue ASTERISK-20442) Reported by: tbsky Tested by: tbsky,alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2141/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f442b40687	Remove dead code and documentation for nonexistent feature. ... multiplelogin was removed from chan_agent back in 1.6.0 when AgentCallbackLogin() was removed. (closes issue AST-948) reported by Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	6f943721f1	Properly handle UAC/UAS roles for SIP session timers ... The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder. This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties. (closes issue AST-922) Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/2118/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b368d7882f	Fix documentation for default username in res_odbc ... This was previously stated to be "root", but is actually the name of the context if unspecified. (closes issue ASTERISK-20258) Reported by: Stefan x git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Darren Sessions	f25975699e	LDAP Realtime Peers Cannot Register ... Prior to 1.8, it was not necessary for an explicit "type" to be set for an asterisk LDAP realtime peer. Now the routine find_peer actually checks the type field during registration and fails to find the peer if it is not set. The attached patches make the realtime type equal whatever type is being searched for if the type is 0 upon return from routine build_peer. (closes issue ASTERISK-17222) Reported by: John Covert Patch by: David Vossel Tested by: Darren Sessions Review: https://reviewboard.asterisk.org/r/2095/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	97b5071653	Fix misleading documentation in agents.conf.sample regarding ackcall usage. ... The documentation made it sound as if the DTMF acknowledgment was needed at the time the agent logs in, rather than when the agent is called. This is likely a relic from the days when there were multiple ways of logging in agents. (closes issue AST-962) reported by Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	80834d503e	Fix incorrectly documented option in queues.conf ... sharedlastcall defaults to "no" not "yes" (closes issue AST-979) reported by Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	80efa31733	Help mitigate potential reinvite glare scenarios. ... When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. For those who are having some deja vu, that's because this patch was originally committed to trunk since there is a new configuration option added. After seeing a bug report about audio being slow to set up on SIP calls, it became apparent that this patch would be the best solution for resolving the issue. The patch is unintrusive and will have no effect unless the option is explicitly enabled. (closes issue AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857) reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	66f69a53c3	Add Digium phones context to sip_notify sample config. ... This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f1bb4eea3d	Update membermacro and membergosub documentation in queues.conf.sample. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e9a0da476a	Add ability to ignore layer 1 alarms for BRI PTMP lines. ... Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	5fcb78840f	Remove a few more files related to chan_usbradio and app_rpt. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	fe1d703804	Set snarkiness = 0 in cdr_adaptive_odbc.conf.sample ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	cc470930ba	Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE ... (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	0106636e42	Don't allow trunkfreq to be greater than 1000ms. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	8357dd7a2d	Don't enable sqlite3 CDRs by default in sample configs. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago