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2008-06-12Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵jpeeler1-14525/+0
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-11Fix pseudo channel allocation errors on startup when using SS7mattf1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121914 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-11Make sure we hangup any calls we have and NULL out the ss7call value when we ↵mattf1-0/+7
get a reset circuit message. Fixes crash bug git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121857 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-04Merged revisions 120425 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120425 | tilghman | 2008-06-04 13:35:47 -0500 (Wed, 04 Jun 2008) | 6 lines If we fail to setup the PRI request channel, don't continue, exit with an error. (closes issue #11989) Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120426 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-29Merged revisions 119071 via svnmerge from tilghman1-3/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) | 7 lines Call waiting tone occurs too often, because it's getting serviced by both subchannels. (closes issue #11354) Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119072 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-29Merged revisions 118953 via svnmerge from tilghman1-16/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines Add some debugging code that ensures that when we do deadlock avoidance, we don't lose the information about how a lock was originally acquired. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118955 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-23Merged revisions 118163 via svnmerge from jpeeler1-5/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) | 1 line Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118164 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causemvanbaak1-12/+12
- make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-21Merged revisions 117582 via svnmerge from jpeeler1-29/+41
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008) | 2 lines Ensure that passed in zt_chan_conf structure is not modified in mkintf. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117658 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-21Merged revisions 117462 via svnmerge from jpeeler1-154/+154
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) | 3 lines Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure. Another commit is following to make sure the zt_chan_conf structure is not modified. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117628 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-19Change the default for the pridialplan parameter to the far more common case oftilghman1-1/+1
'unknown', and better document the use of each parameter. (closes issue #12633) Reported by: tzafrir Patches: pridialplan_unknown_2.diff uploaded by tzafrir (license 46) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117182 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-16Try to see if we can make our ringback situation a little bettermattf1-4/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116797 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-13Need to clear calling_party_cat variable after we retrieve itmattf1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115941 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-13Add support for receiving calling party categorymattf1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115939 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-11Add Zap MTP2 support to chan_zapmattf1-2/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115600 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-11Open up audio channel when we get ACM on SS7 eventmattf1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115598 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-08Remove unused code as well as demote an error message to a debug messagemattf1-6/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115548 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Add two new console commands "pri show version" and "ss7 show version" that ↵bbryant1-0/+32
will show the version of each library respectively. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115078 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Allow dringXrange to properly default to 10, as was done in 1.4.qwell1-10/+9
dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified. (closes issue #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded by qwell (license 4) Tested by: bjm git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114922 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-28Fix deadlock issue in chan_zap with libss7 due to channel variables being ↵mattf1-1/+11
set with the channel pvt lock being held. #12512 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114776 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-24Pass the hangup cause all the way to the calling app/channel.mvanbaak1-2/+2
(closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114637 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21(closes issue #6113)jpeeler1-1/+16
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21Add support for generic name transmission (#12484) on SS7 in chan_zapmattf1-5/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114389 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-19SS7:Added - Generic Name / Access Transport / Redirecting Number handling. ↵mattf1-3/+42
#12425 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114303 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-18Merged revisions 114257 via svnmerge from mmichelson1-4/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines Clearing up error messages so they make a bit more sense. Also removing a redundant error message. Issue AST-15 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114259 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-16Merged revisions 114184 via svnmerge from kpfleming1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114185 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-12Make sure linkset is locked exiting ss7_start_callmattf1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114093 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-12Make sure we start incoming calls on SS7 with echo cancellation enabled. ↵mattf1-4/+14
Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114092 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-03Merged revisions 112599 via svnmerge from mmichelson1-16/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines Fix the testing of the "res" variable so that it is more logically correct and makes the correct warning and debug messages print. (closes issue #12361) Reported by: one47 Patches: chan_zap_deferred_digit.patch uploaded by one47 (license 23) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@112600 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-01This adds DNS SRV record support to DNS manager. If there is a SRV record ↵jpeeler1-2/+0
for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@112207 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-01Now that zaptel trunk has been removed, add the PSTN deprecation notice to ↵russell1-0/+2
chan_zap, as well. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@112124 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-26Large cleanup of DSP codeqwell1-16/+15
Per comments from dimas: 1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones. 2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before. 3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best. Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too. 4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used. 5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel. This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality. (closes issue #11968) Reported by: dimas Patches: v2-11968-dsp.patch uploaded by dimas (license 88) v4-11968-zap.patch uploaded by dimas (license 88) Tested by: dimas, qwell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@111022 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-19Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.qwell1-4/+4
This was very misleading. Early cleanup for issue #11968 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110161 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-19Rename very poorly named function to reflect what it actually does. This ↵qwell1-12/+12
was causing quite a bit of confusion for me... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110132 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18Fix a typo which caused a double free in chan_zap. This was discoveredmmichelson1-1/+1
by Juggie while attempting to load chan_zap. Apparently this would happen if an error were encountered while trying to process zapata.conf. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@109802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-12add support for named sections in zapata.conf, and fix a few bugs in config ↵kpfleming1-18/+63
file parsing (closes issue #9503) Reported by: tzafrir Patches: fix_cleanups uploaded by tzafrir (license 46) zapata_sections uploaded by tzafrir (license 46) skipchannel_options uploaded by tzafrir (license 46) conf_sample uploaded by tzafrir (license 46) patches updated by me to better conform to coding guidelines and fix some problems git-svn-id: http://svn.digium.com/svn/asterisk/trunk@108286 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10Merged revisions 107173 via svnmerge from qwell1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | 5 lines Make sure to reenable echo can after a "failed" (canceled, etc) three-way call. (closes issue #11335) Reported by: rebuild ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107177 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-08Merged revisions 106945 via svnmerge from kpfleming1-5/+11
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down" ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106946 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Make sure we don't start a call when we have already done so in response to ↵mattf1-3/+5
a COT message git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106892 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05Merged revisions 106235 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106239 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05Merged revisions 106038 via svnmerge from kpfleming1-16/+23
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one (closes issue #11917) Reported by: mavetju Tested by: mavetju ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106040 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-04Fix minor misuses of snprintftilghman1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105841 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03Fix some code that was improperly changed in revision 104866 from issue #12079.russell1-1/+1
(closes issue #12129, reported by elguero, patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105574 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27reduce indentation in alloc_subrussell1-25/+28
(issue #12079) Reported by: tzafrir Patches: alloc_sub uploaded by tzafrir (license 46) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104866 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-26Merged revisions 104119 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines Merge changes from team/russell/smdi-1.4 This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104120 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-25Deprecate the "stripmsd" option in favor of dialplan substring variable syntax.russell1-0/+1
(closes issue #12060) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104110 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-22Add protection to chan_zap build when NEONMWI events are not defineddbailey1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104045 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-21Added configuration distinction between neon and fsk mwi detectiondbailey1-32/+77
Add the detection for neon MWI events got rid of extraneous handle_init_event call in monitor loop git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104024 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-20Merged revisions 103953 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6 lines Don't wait for additional digits when overlap dialing is enabled if the setup message contains the sending_complete information element. (closes issue #11785) Reported by: klaus3000 Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103954 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-19(closes issue #11864)jpeeler1-0/+3
Reported by: julianjm Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm (license 99) Patch fixes problem of device state incorrectly reporting idle before PBX answers incoming call on FXO channel. Device status is updated now during new channel creation. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103818 f38db490-d61c-443f-a65b-d21fe96a405b
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