asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	bcdf507f3c	Change D-channel warning to be less confusing on non-NFAS setups. The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8711d897d0	Make duplicate call ptr warning message more helpful. * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f8b799c0c1	Made ISDN not add numbering plan prefix strings to empty numbers. When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9eb7ccef76	Rework sig_pri_hangup() to be simpler and clearer. JIRA AST-675 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7dc49195d8	Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9328590ddb	Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c4afd498c0	Merged revisions 330033 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r330033 \| rmudgett \| 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) \| 15 lines Datacalls with B410P fail. Incoming and outgoing call legs of a data call are using different formats: a-law, u-law. When the call is bridged, the media stream is run through translation to convert the media formats. The translation is bad for data calls. * Make incoming call that does not explicitly specify u-law or a-law use the DAHDI channel's default law. The outgoing call always uses the default law from the DAHDI channel. (closes issue ABE-2800) Patches: jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a7394bcd28	Backport useful CLI "pri show channels" command to v1.8. The "pri show channels" command is useful for debuging to see if there are any stuck B channels. .......... r307964 \| rmudgett \| 2011-02-15 15:42:55 -0600 (Tue, 15 Feb 2011) \| 9 lines Add CLI "pri show channels" command. List the current mapping of DAHDI B channels to Asterisk channel names and which calls are on hold or call-waiting. Calls on hold or call-waiting are not associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 .......... r308205 \| rmudgett \| 2011-02-17 14:21:56 -0600 (Thu, 17 Feb 2011) \| 1 line Add more verbage to CLI command 'pri show channels' usage. .......... r312579 \| rmudgett \| 2011-04-04 11:17:58 -0500 (Mon, 04 Apr 2011) \| 59 lines Change also updates 'pri show channels' command with the "chan idle" column to report if a channel is available for use. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	209d8d3c15	PRI early media won't ring. And another way to pass early media. Don't indicate that there is inband information present, just assume that the B channel is connected. * Restore clearing the dialing flag Rx squelch unconditionally when a PROCEEDING message comes in. (closes issue #19268) Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett (license 664) Tested by: tbsky git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0ec0f72506	Unable to pickup DAHDI/PRI call because call state is reported as DIALING. The channel state is not updated to RINGING when an ALERTING message is received. Regression caused when sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added missing channel state update to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes issue #19257) Reported by: alecdavis Patches: issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	90c544c0e1	Don't get early media for ISDN on outgoing calls. It looks to be a long-standing misinterpretation of the progress indicator ie values: 1 - Call is not end-to-end ISDN; further call progress information may be available in-band. 8 - In-band information or an appropriate pattern is now available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled as early media probably because the meaning of the second half of it's description was overlooked. * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. (closes issue #18868) Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: satish_lx .......... No inband progress on PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider sends an ALERTING with "Inband information or appropriate pattern now available", but Asterisk only generates and passes the RING to the SIP extension, not the inband message. Unfortunately, the inband message is not a ringback tone but a prompt that says the number is not in service. The SIP extension then hears two rings and the call is hungup which confuses the caller. * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband audio is indicated with an ALERTING message. (closes issue #19246) Reported by: cristiandimache Patches: issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: cristiandimache git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c5ad2f12a0	The dahdi_hangup() call does not clean up the channel fully. After dahdi_hangup() has supposedly hungup an ISDN channel there is still traffic on the S0-bus because the channel was not cleaned up fully. Shuffled the hangup code to include some missing cleanup. Also fixed some code formatting in the area. I think the primary missing clean up code was the call to tone_zone_play_tone() to turn off any active tones on the channel. (closes issue #19188) Reported by: jg1234 Patches: issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested by: jg1234 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	20ef1e9c95	Fix ISDN calling subaddr User Specified Odd/Even Flag Calculation of the Odd/Even flag was wrong. Implement correct algo, and set odd/even=0 if data would be truncated. Only allow automatic calculation of the O/E flag, don't let dialplan influence. (closes issue #19062) Reported by: festr Patches: bug19062.diff2.txt uploaded by alecdavis (license 585) Tested by: festr, alecdavis, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4242cb82f4	Crash if ISDN span layer 1 is down on initial load. Regression from -r312575 B channel shifting during negotiation. * Also combine updating the alarm flag with clearing the resetting flag. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	458a57d1d3	Merged revisions 312574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312574 \| rmudgett \| 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) \| 45 lines Merged revisions 312573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312573 \| rmudgett \| 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) \| 38 lines Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4f3cf039f4	Race condition when ISDN CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY could sometimes be processed before the call_forward dial string is recognized. * Moved setting the call_forwarding dial string after sending a response to the initiator and just queue an empty frame to wake up the media thread instead of an AST_CONTROL_BUSY. * Added check for empty rerouting/deflection number and respond with an error. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8bfde13607	Make pri parameter description consistent. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5b9f9f78ca	Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7b353a26ae	sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails. (closes issue #18874) Reported by: cmaj Patches: patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830) JIRA SWP-3172 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a5f6367057	No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	39728fbe4d	Merged revisions 305342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305342 \| rmudgett \| 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) \| 14 lines Merged revisions 305341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305341 \| rmudgett \| 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) \| 7 lines Obtain the pri lock for PRI queue counters. Need to obtain the pri lock when calling pri_dump_info_str() to avoid a reentrancy problem when calculating the Q.921 Q count statistic. JIRA AST-484 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8e51d30b67	Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 \| rmudgett \| 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) \| 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 \| rmudgett \| 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) \| 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	78e1319a13	Deadlock between dahdi_request() and pri_dchannel() processing an incomming call. The sig_pri_new_ast_channel() is called with the channel private lock held when pri_dchannel() calls it and no channel private lock held when dahdi_request() calls it. The use of pri_grab() in sig_pri_new_ast_channel() could leave the channel private lock held when it returns if the lock was not held before calling it. Make sig_pri_new_ast_channel() just lock the PRI span lock instead of using pri_grab(). It is safe to do this because dahdi_request() does not have the channel private lock and the deadlock potential with the PRI span lock is only between pri_dchannel() and other threads. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@301946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	2baf7ac892	Merged revision 300711 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r300711 \| rmudgett \| 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) \| 14 lines A call retrieved from hold may wind up with no audio. If the retrieved call is natively bridged then the call may not have any audio path. The following warning message is given: "Failed to add <dfd> to conference <chan>/<chan>: Invalid argument". * Open the media on a B channel when pri_fixup_principle() moves the call from a no_b_channel channel to a real channel. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@300714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	30a74dd9fe	Chan_dahdi sends an empty COLP on the bridged channel. Chan_dahdi always inserts a connected party IE when you call from one dahdi channel to another dahdi channel, even if no such information was received on the 2nd channel. This clears the display of many phones. * Removed leftover artifact from before the valid flag was added. * Updated all of the channel's caller id information with the new connected line information instead of just the string parts. (closes issue #18508) Reported by: wimpy Patches: issue18508_trunk.patch uploaded by rmudgett (license 664) Tested by: wimpy, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@299405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3ed89f0e89	Merged revisions 298194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298194 \| rmudgett \| 2010-12-13 11:04:41 -0600 (Mon, 13 Dec 2010) \| 26 lines Merged revisions 298193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298193 \| rmudgett \| 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) \| 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered transfers. Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING message is not received. The debug output shows that the DTMF begin event is seen, but the DTMF end event is missing. When the DTMF begin happens, the call is muted so we now have one way audio (until a DTMF end event is somehow seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin and DTMF end events if we are overlap dialing and have not seen a PROCEEDING message. * Added a debug message when absorbing a DTMF event. JIRA SWP-2690 JIRA ABE-2697 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@298195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ac9434aecc	Merged revisions 294822 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294822 \| rmudgett \| 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) \| 18 lines Merged revisions 294821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 \| rmudgett \| 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) \| 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3f9644b7db	Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	70415ccdfd	No need to define the struct if there are no users. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d411ade931	Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	d1aceac167	Several more defines that need to be altered for compiling against an older version of libpri git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	cdc420235e	Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	eeec27dafc	Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call. When a call is transfered by ECT or implicitly by disconnect in sig_pri or implicitly by disconnect in chan_misdn, the connected line information is not exchanged. The connected line interception macros also need to be executed if defined. The CALLER interception macro is executed for the held call. The CALLEE interception macro is executed for the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: abe_2589_c3bier.patch uploaded by rmudgett (license 664) abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/958/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7e28c7c90d	Send CONNECT_ACKNOWLEDGE for CIS calls too. The originator of the Q.SIG call completion signaling link was not changed to the active state when the CONNECT message came in. The T309 processing would immediately kill the signaling link because it was not in the active state. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cb82f20163	Avoid deadlock processing incoming AOC-E messages. Deadlock avoidance for the owner channel was not done when processing incoming AOC-E messages. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bb8e2fcef8	Revert stuff not ready for commit in -r289054. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	34b3615fff	Break up long ast_manager_event_multichan() event lines. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c54833f35d	Merged revisions 286116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286116 \| rmudgett \| 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) \| 18 lines Merged revisions 286113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 \| rmudgett \| 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) \| 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	5f2b40c699	Simplified pri_dchannel() poll timeout duration code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c271acacda	Made output libpri event names if pri debugging is enabled when sig_pri processes them. * Simplified CLI "pri debug xx span xx" command code and removed redundant debugging enabled messages. * Made CLI "pri debug xx span xx" command only close the debugging log file if it was opened. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c453d72423	Merged revisions 283049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283049 \| rmudgett \| 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) \| 29 lines Merged revisions 283048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283048 \| rmudgett \| 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) \| 22 lines Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8)" (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	4aed988d66	Merged revisions 282607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282607 \| tilghman \| 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) \| 9 lines Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes. (closes issue #16770) Reported by: jamicque Patches: 20100413__issue16770.diff.txt uploaded by tilghman (license 14) 20100811__issue16770.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	e32da6f7a5	Fix compilation error in chan_dahdi (strdupa -> ast_strdupa). (closes issue #17751) Reported by: b11d Patches: strdupa_oops.diff uploaded by malcolmd (license 924) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	e7b5069c9f	Fix inband DTMF detection on outgoing ISDN calls. This is a regression from the sig_pri split from chan_dahdi. When a call is first initiated, the inband DTMF detector is not enabled if it's an outgoing ISDN call. However, it needs to be turned on once the media path starts up. This handling was put back in the open_media() callback of chan_dahdi. In sig_pri, open_media() calls were added to a few places where it was needed, including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING. Thanks to rmudgett for helping me with the patch! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	301505c4c4	Rename sig_pri_pri to sig_pri_span. More descriptive of concept. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7066a7f233	Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char tag; }; struct ast_party_dialed { struct { char str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	30888f913d	Merged revisions 272446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272446 \| rmudgett \| 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) \| 10 lines ss_thread calls pri_grab without lock during overlap dial Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	93a5e74e37	Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn. Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn. Review: https://reviewboard.asterisk.org/r/696/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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118 Commits (bcdf507f3c67ad8c01ef1d989ce53e45bea0f201)