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140 Branches
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7516 Commits (28464199ee83b765d2d68b7bc013d17f5a1bb7f4)
 

Author SHA1 Message Date
Joshua Colp 28464199ee Bring RTP back to Asterisk at the end of a native bridge no matter what.
18 years ago
Christian Richter 37ded96cfa if the bridged partner is mISDN too we should not send dtmf tones, they are transmitted inband always
18 years ago
Christian Richter 7bb272f942 if we have already some digits, we just stop the tones.
18 years ago
Christian Richter 9809905c76 added check for NULL Pointer when calling misdn_new. Asterisk does not allow us to create channels anymore when stop gracefully is used :). also modified the restart_indicator to 0
18 years ago
Joshua Colp 41421be994 Improve deadlock handling of the channel list. (issue #8376 reported by one47)
18 years ago
Christian Richter 5cc2b1078e fixed problem that the dummybc chanels had no lock, checking for the lock now. Also fixed the channel restart stuff, we can now specify and restart particular channels too.
18 years ago
Russell Bryant 086aba207e Don't automatically hang up after running Dictate so that callers can exit
18 years ago
Joshua Colp d8fb68b9cb Don't call ast_waitstream_full when the control file descriptor and audio file descriptor are not set, simply call ast_waitstream! (issue #8530 reported by rickead2000)
18 years ago
Joshua Colp 49ffa786ff Do a DNS lookup immediately upon calling the dnsmgr function, don't wait until a refresh happens. (issue #9097 reported by plack)
18 years ago
Russell Bryant c00062ff31 Fix a problem where saying a character wouldn't properly break out when the caller pressed '#'
18 years ago
Jason Parker a3961d646a Don't try to save voicemail greetings unless the user presses '1' to accept/save.
18 years ago
Joshua Colp 9c0627eab0 Allow the 'g' option to work if used with the 'S' option. (issue #9888 reported by gasparz)
18 years ago
Joshua Colp 084ede4507 Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia)
18 years ago
Russell Bryant 3a2e0e1ed9 We have some bug reports showing crashes due to a double free of a channel.
18 years ago
Joshua Colp 76fdb9418b Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz)
18 years ago
Joshua Colp a05ab92728 Revert channel name splitting fix for Zap. The moral of the story is don't use - in your user/peer names. (issue #9668 reported by stevedavies)
18 years ago
Christian Richter f002ad09a3 briding is a bool, fixed copy and paste issue.
18 years ago
Christian Richter e7590d0aec simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination.
18 years ago
Nadi Sarrar e0f4f4969c Backport of the overlap_dial functionality from asterisk-1.4's chan_misdn.
18 years ago
Christian Richter 3cd1c84e8d added possibility to deactivate bridging per port
18 years ago
Tilghman Lesher dd412388a1 According to MATH, 0+1181000386 = 1181000448. Oops.
18 years ago
Tilghman Lesher c78acd2896 Add revision Id tags (by request of tzafrir)
18 years ago
Joshua Colp 22fe1b73cc It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)
18 years ago
Tilghman Lesher c0ce087e43 Issue 9818 - Fix for issue 8329 breaks pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime, but only because we lack core API to do it.
18 years ago
Tilghman Lesher fc6d28a932 If the value of a variable passed to FIELDQTY is blank, then FIELDQTY should return 0, not 1.
18 years ago
Olle Johansson c4e7d9fef5 Issue #9802 - Change inuse counter on CANCEL
18 years ago
Tilghman Lesher 2cb2558eb1 Issue 9791 - Fix pronunciation of seconds in Dutch
18 years ago
Joshua Colp ad2f350d39 Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)
18 years ago
Christian Richter 17175c7d54 we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example.
18 years ago
Kevin P. Fleming cba8e2f704 ensure that variables are set on a newly created channel before we start a PBX on it
18 years ago
Kevin P. Fleming 9edd1e094c if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it
18 years ago
Russell Bryant 2f0f1f5e00 Revert revision 62417 as someone reported problems with it to Mark. This was
18 years ago
Russell Bryant 03ad135134 Fix a memory leak that I just noticed in the device state handling in app_queue.
18 years ago
Christian Richter 0b6da8d56e we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
18 years ago
Steve Murphy fd1fc0a9c1 This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
18 years ago
Olle Johansson 86882515a8 Not getting an ACK to a 200 OK in the initial invite is critical to the call.
18 years ago
Olle Johansson 21ea4dc3f1 Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
18 years ago
Christian Richter 58bcd919d5 fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
18 years ago
Christian Richter 06b2955d26 we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested.
18 years ago
Tilghman Lesher b2f84c347f How is it that we never caught that this is returning the opposite of our documentation, until now?
18 years ago
Jason Parker dfd24c33d2 If we have a negative current message, we shouldn't go back even further...
18 years ago
Olle Johansson 9ebfde54a1 Fixing possible bug in auth of BYE
18 years ago
Olle Johansson 80e4abca3d Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!)
18 years ago
Olle Johansson aa9ff74af5 Issue #9726 - rlister - Better logging for ACL denials
18 years ago
Christian Richter b60fd4bc20 in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
18 years ago
Joshua Colp 3b1ad79633 Only perform stripping of - strings from the channel name for Zap channels. Anywhere else we might remove a legitimate part of a device name. (issue #9668 reported by stevedavies)
18 years ago
Tilghman Lesher 29aa7c809b Issue 9121 - fixups for safe_asterisk script
18 years ago
Jason Parker 074cc21291 Fix an issue with trying to kill a thread before it gets created.
18 years ago
Olle Johansson 07ba0e379b Do not allocate SIP pvt's for PEERs we can not reach.
18 years ago
Matthew Fredrickson 818c25352e Make sure we only create a DSP if it's requested on SUB_REAL
18 years ago