950e756132
Make ringing clearer
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3391 f38db490-d61c-443f-a65b-d21fe96a405b
2004-07-08 09:20:14 +00:00
974652e0e7
Cleanups to the ordering of events in dial, don't freak out on the wrong codec
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3385 f38db490-d61c-443f-a65b-d21fe96a405b
2004-07-07 16:02:13 +00:00
96c27903b1
Add outgoing OSP support (SIP only at this point)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3296 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-25 03:59:07 +00:00
1d287c5c28
Set ${DIALSTATUS} from app_dial, make zap return BUSY on a phone that is busy, update macro-stdexten to demo
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3282 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-23 03:16:58 +00:00
31e064cf03
Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h).
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3277 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-22 19:32:52 +00:00
ae5c80e1de
Merge major BSD mutex and symbol conflict patches (bug #1816 ) (link patch still pending)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3273 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-22 17:42:14 +00:00
abda2d4bdf
Make sure outgoing is not NULL
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3270 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-22 13:53:45 +00:00
d97e97521f
Don't stop moh until after announcement finishes (bug #789 )
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3262 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-21 18:28:35 +00:00
92a78593cb
Fix app_dial buglet
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3256 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-21 13:30:58 +00:00
ff02c2b632
I don't know what this last update was for but app_dial.c was broken
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so I added these 2 variable defs to make it compile right to spare
all the end users...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3255 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-21 12:55:53 +00:00
172f7c07c3
Add promiscuous redirect option
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3254 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-21 06:11:56 +00:00
da48cd7fcf
Typo in comment in app_dial.c
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3244 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-20 06:24:25 +00:00
81151ce938
Merge Matt's work
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3204 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-14 19:43:41 +00:00
c15a46e7ea
fix D() issue with to short of a duration
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3152 f38db490-d61c-443f-a65b-d21fe96a405b
2004-06-06 02:30:10 +00:00
43a79d265d
Copy hangup cause at end of a good call
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3042 f38db490-d61c-443f-a65b-d21fe96a405b
2004-05-22 23:17:33 +00:00
abc625c7e8
Merge Mike's announcement enhancements
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3020 f38db490-d61c-443f-a65b-d21fe96a405b
2004-05-20 00:29:09 +00:00
fae3002618
Clarify app_dial documentation regarding transfers
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3006 f38db490-d61c-443f-a65b-d21fe96a405b
2004-05-18 18:27:55 +00:00
52ae3f42d4
Code cleanup
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2919 f38db490-d61c-443f-a65b-d21fe96a405b
2004-05-07 21:14:55 +00:00
c145725240
Created API call ast_dtmf_stream
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int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digits,int between)
changed app_senddtmf.c to use this new call
added D() parameter to app_dial to allow post connect dtmf stream to be sent using above call
-Tony
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2918 f38db490-d61c-443f-a65b-d21fe96a405b
2004-05-07 20:39:14 +00:00
5d7c4bb03e
strlen fixes and don't destroy SIP channels that still have pending things on them (bug #1552 et al)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2890 f38db490-d61c-443f-a65b-d21fe96a405b
2004-05-04 19:11:25 +00:00
5897660860
Bug Fix
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2784 f38db490-d61c-443f-a65b-d21fe96a405b
2004-04-27 16:42:43 +00:00
aa820ac940
applied final release of bug 1353 per Mark's permission
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2782 f38db490-d61c-443f-a65b-d21fe96a405b
2004-04-26 23:22:34 +00:00
e22999aa42
Make app_dial complain if the timeout passed is non numeric
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2601 f38db490-d61c-443f-a65b-d21fe96a405b
2004-04-02 07:47:23 +00:00
def55d9695
Remember when we started/stopped ringing so that ringing twice works as expected
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2462 f38db490-d61c-443f-a65b-d21fe96a405b
2004-03-19 00:08:43 +00:00
759ff4c35f
Make RDNIS really work right
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2456 f38db490-d61c-443f-a65b-d21fe96a405b
2004-03-18 20:02:15 +00:00
096423e264
Populate RDNIS information
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2452 f38db490-d61c-443f-a65b-d21fe96a405b
2004-03-18 05:48:40 +00:00
abd154ee5e
Add absolute timeout option to dial that applies when call is up (bug #1107 )
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2357 f38db490-d61c-443f-a65b-d21fe96a405b
2004-03-06 00:02:40 +00:00
f0a93508bf
Check result of malloc in app_dial.c
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2255 f38db490-d61c-443f-a65b-d21fe96a405b
2004-02-27 04:24:42 +00:00
f201de8329
Fix another typo in the app_dial description
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2211 f38db490-d61c-443f-a65b-d21fe96a405b
2004-02-22 05:31:42 +00:00
bfe8957e96
Spelling error in app_dial.c
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2177 f38db490-d61c-443f-a65b-d21fe96a405b
2004-02-15 08:33:19 +00:00
173af85825
A little adjustment to the previous one (passing the clid&ani on call_forward)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2044 f38db490-d61c-443f-a65b-d21fe96a405b
2004-01-15 23:38:23 +00:00
b362c2a04b
Pass the callerid & ani to the new b-leg if the current b-leg requests call forwarding
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2043 f38db490-d61c-443f-a65b-d21fe96a405b
2004-01-15 23:32:27 +00:00
19db091314
X out announce stuff (bug #760 )
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2039 f38db490-d61c-443f-a65b-d21fe96a405b
2004-01-15 20:20:21 +00:00
004e6fbc01
Allow doing digital PRI to PRI calls automatically
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1868 f38db490-d61c-443f-a65b-d21fe96a405b
2003-12-18 23:42:10 +00:00
a0f138dbe1
fix minor typos in descrip. Bug#635
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1844 f38db490-d61c-443f-a65b-d21fe96a405b
2003-12-09 19:07:45 +00:00
85e718e4d9
Forward language choice
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1826 f38db490-d61c-443f-a65b-d21fe96a405b
2003-12-04 22:26:38 +00:00
6f7fbed695
I have no idea what the heck that was supposed to be
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1821 f38db490-d61c-443f-a65b-d21fe96a405b
2003-12-03 15:21:58 +00:00
c8c98c2ede
Got rid of un-necessary 'c' and 'd' options in app_dial.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1804 f38db490-d61c-443f-a65b-d21fe96a405b
2003-11-28 04:38:07 +00:00
ab55ec20e6
minor language tweak. Thanks Carp
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1700 f38db490-d61c-443f-a65b-d21fe96a405b
2003-11-06 04:08:40 +00:00
76c431dd65
WHen making compatible for bridge, stop generators, etc.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1697 f38db490-d61c-443f-a65b-d21fe96a405b
2003-11-05 20:53:49 +00:00
9564ebe2cd
Add announce feature to dial
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1657 f38db490-d61c-443f-a65b-d21fe96a405b
2003-10-22 04:25:13 +00:00
143f06cec4
Add 486 message (SIP) to 17 cause (PRI) translation when you call Dial app and then Hangup:
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exten => _X.,1,Dial(SIP/user)
exten => _X.,2,Hangup()
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1596 f38db490-d61c-443f-a65b-d21fe96a405b
2003-10-01 16:05:40 +00:00
cfa5d5657f
Add usecallingpres keyword to zapata.conf for PRI to PRI call so that we would take the presentation mode and send it to outbound call.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1595 f38db490-d61c-443f-a65b-d21fe96a405b
2003-10-01 15:59:26 +00:00
949a70a88e
Copy musiconhold
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1479 f38db490-d61c-443f-a65b-d21fe96a405b
2003-09-06 19:28:52 +00:00
72200fa226
Add restrictcid=[yes|no] for chan_zap and chan_sip. You can specify if the sip/zap users will have the callerid transmitted as ANI or CALLERID.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1476 f38db490-d61c-443f-a65b-d21fe96a405b
2003-09-05 04:00:57 +00:00
a5c52defec
Allow "g" option in dial to go to next extension (bug #189 )
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1445 f38db490-d61c-443f-a65b-d21fe96a405b
2003-08-28 16:42:15 +00:00
24183cf0a7
Save the dstchannel before the call gets answered so that we would know what destination channel was dialed when we get the channel hanged up
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1331 f38db490-d61c-443f-a65b-d21fe96a405b
2003-08-14 20:48:44 +00:00
fab8b0d20a
Add alternate queueing strategies. Implment ringall, roundrobin, and random
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1238 f38db490-d61c-443f-a65b-d21fe96a405b
2003-07-30 16:10:51 +00:00
3d5784cb8e
Make sure o->chan is non-NULL
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1222 f38db490-d61c-443f-a65b-d21fe96a405b
2003-07-25 15:32:02 +00:00
5d4af0b9c4
Fix the compile problem
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1202 f38db490-d61c-443f-a65b-d21fe96a405b
2003-07-22 23:08:45 +00:00
markster