Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation.  However, if you used it, it required using different
functions for modifying scheduler contents.  This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there.  This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.

In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.

Review: https://reviewboard.asterisk.org/r/1007/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r297957 | dvossel | 2010-12-09 15:32:20 -0600 (Thu, 09 Dec 2010) | 11 lines
  
  Fixes issue with outbound google voice calls not working.
  
  Thanks to az1234 and nevermind_quack for their input in helping debug the issue.
  
  (closes issue #18412)
  Reported by: nevermind_quack
  Patches:
        fix uploaded by dvossel (license 671)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/984/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r291827 | dvossel | 2010-10-14 16:27:42 -0500 (Thu, 14 Oct 2010) | 18 lines
  
  Safer xml parsing, treat all clients the same, and better local candidate selection.
  
  The gtalk channel driver was doing several unsafe operations
  in regards to how it parsed incoming XML messages.  I have cleaned
  that code up so it should be much safer now.
  
  We now treat all clients types the same.  We have no reason to
  distinguish between GMAIL and GOOGLE VOICE clients anymore because
  they all work the same way.
  
  I also modified how the local ip is found.  If no bindaddress is provided
  in the config file, we attempt to determine the local ip we
  would use to connect to google.com.  If that fails, then
  we fall back to the ast_find_ourip() function as a last resort.
  Using the new method makes it much less likely that we would ever
  advertise a local RTP candidate as a loopback address.
........


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https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) | 11 lines
  
  Add the ability for ast_find_ourip to return IPv4, IPv6 or both.
  
  While testing chan_gtalk I noticed jabber was using my IPv6 address
  and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip()
  to return both IPv6 and IPv4 results.  Adding a family parameter gives you
  the ablility to choose.
  
  Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results.
  
  Review: https://reviewboard.asterisk.org/r/973/
........


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https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r291578 | dvossel | 2010-10-13 17:46:34 -0500 (Wed, 13 Oct 2010) | 4 lines
  
  More fixup for chan_gtalk.
  
  This patch makes the xml parsing safer.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291579 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) | 19 lines
  
  Gtalk enhancements and general code cleanup.
  
  This patch includes several chan_gtalk enhancements.
  Two new gtalk.conf options have been added, externip
  and stunadd.  Setting externip allows us to
  manually specify what the external IP address is
  outside of a NAT environment.  Setting the stunaddr
  option to a valid stun server allows for that external
  ip to be retrieved via a STUN server automatically.  This
  external IP is then advertised during call setup as
  a possible candidate.
  
  I have also attempted to clean up chan_gtalk's code
  so it meets our coding guidelines. During this cleanup
  I noticed several things that need to be done in the
  code and made a TODO section at the top of the file.
........


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https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r290973 | dvossel | 2010-10-08 15:44:59 -0500 (Fri, 08 Oct 2010) | 12 lines
  
  Make outbound Google Voice calls.
  
  This patch allows for outbound Google Voice calls to be
  dialed from Asterisk using chan_gtalk. Below is an example
  dialstring.
  
  exten -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,)
  
  In this example, 'asterisk' is the jabber.conf profile configured
  to connect to your gmail account. In order to receive Google Voice
  calls make sure to enable 'allowguest=yes' in gtalk.conf.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r290829 | dvossel | 2010-10-07 17:38:05 -0500 (Thu, 07 Oct 2010) | 6 lines
  
  Add Philippe Sultan to chan_gtalk author list.
  
  Philippe has made some notable contributions to the
  gtalk channel driver.  His name deserves to be listed
  amoung the authors of that file.  Thanks Philippe!
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r290828 | dvossel | 2010-10-07 16:44:58 -0500 (Thu, 07 Oct 2010) | 5 lines
  
  Outbound gtalk calls now work correctly.
  
  There was a problem with how the candidates were being
  built on an outbound call. This patch fixes that.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r290674 | dvossel | 2010-10-06 16:22:51 -0500 (Wed, 06 Oct 2010) | 4 lines
  
  Fixes commented out code to use #if 0 instead.
  
  Thanks to rmudgett for catching this!
........


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https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06 Oct 2010) | 12 lines
  
  Fixes gtalk outbound DTMF to work properly.
  
  Outbound DTMF with gtalk needs to be done within the RTP stream.  I discovered
  this after investigating a packet capture from the gmail client.  Instead of
  performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive
  on the RTP stream using RFC2833 way of doing things.  Chan_gtalk also had an issue
  with negotiating RTP payload type 106 for the telephony-event and then sending
  DTMF as payload 101.  This has been resolved by always negotiating 101 as the payload
  type like we do everywhere else.  With this patch, incoming google voice calls forwarded
  to Asterisk via gtalk work.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290649 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) | 6 lines
  
  Fixes chan_gtalk to work with gmail client
  
  This patch was written by Philippe Sultan (phsultan). Thanks
  for keeping this up to date!
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290480 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.

Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.

(closes issue #17565)
Reported by: russell
Patches: 
      asteriskv6-test-report.pdf uploaded by russell (license 2)

Review: https://reviewboard.asterisk.org/r/743



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default.  This value is required
in order for the adaptive jitterbuffer to work correctly.  To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3

JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).

(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo

Review: https://reviewboard.asterisk.org/r/88/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes a crash reported in #asterisk-dev where chan_mgcp unexpectedly
allocated an RTP instance from res_rtp_multicast, since by not specifying an
engine, you get the first one in the list of engines.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211732 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CEL is the new system for logging channel events.  This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records.  For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.

Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code.  Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.

Review: https://reviewboard.asterisk.org/r/239/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3

invalid bindaddr was specified in gtalk.conf.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196381 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There is a lot that could be said about this, but the patch is a big 
improvement for performance, stability, code maintainability, 
and ease of future code development.

The channel list is no longer an unsorted linked list.  The main container 
for channels is an astobj2 hash table.  All of the code related to searching 
for channels or iterating active channels has been rewritten.  Let n be 
the number of active channels.  Iterating the channel list has gone from 
O(n^2) to O(n).  Searching for a channel by name went from O(n) to O(1).  
Searching for a channel by extension is still O(n), but uses a new method 
for doing so, which is more efficient.

The ast_channel object is now a reference counted object.  The benefits 
here are plentiful.  Some benefits directly related to issues in the 
previous code include:

1) When threads other than the channel thread owning a channel wanted 
   access to a channel, it had to hold the lock on it to ensure that it didn't 
   go away.  This is no longer a requirement.  Holding a reference is 
   sufficient.

2) There are places that now require less dealing with channel locks.

3) There are places where channel locks are held for much shorter periods 
   of time.

4) There are places where dealing with more than one channel at a time becomes 
   _MUCH_ easier.  ChanSpy is a great example of this.  Writing code in the 
   future that deals with multiple channels will be much easier.

Some additional information regarding channel locking and reference count 
handling can be found in channel.h, where a new section has been added that 
discusses some of the rules associated with it.

Mark Michelson also assisted with the development of this patch.  He did the 
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it 
much easier to deal with holding on to a channel pointer for an extended period 
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.

Thanks to David Vossel for his assistance with this branch, as well.  David 
did the conversion of the DAHDIScan application by making it become a wrapper 
for ChanSpy internally.

The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.

Review: http://reviewboard.digium.com/r/203/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187772 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines
  
  Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces
  
  To drill into the xmpp to find the capabilities between channels, chan_gtalk 
  calls iks_child() and iks_next(). iks_child() and iks_next() are functions in 
  the iksemel xml parsing library that traverse xml nodes. The bug here is that 
  both iks_child() and iks_next() will return the next iks_struct node 
  *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, 
  which in most cases, it is, but in this case (a call being made from the 
  Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data 
  (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, 
  so capabilities don't match, and a call cannot be made.
  
  iks_first_tag() and iks_next_tag(), on the other hand, will not return the 
  very next iks_struct, but will check to see if the next iks_struct is of 
  type IKS_TAG. If it isn't, it will be skipped, and the next struct of type 
  IKS_TAG it finds will be returned. This assures that chan_gtalk will find 
  the iks_struct it is looking for.
  
  This fix simply changes all calls to iks_child() and iks_next() to become 
  calls to iks_first_tag() and iks_next_tag(), which resolves the capability 
  matching.
  
  The following is a payload listing from Empathy, which, due to the extraneous 
  whitespace, will not be parsed correctly by iksemel:
  
  <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/>
   <payload-type clockrate='8000' name='PCMA' id='8'/>
   <payload-type clockrate='8000' name='PCMU' id='0'/>
   <payload-type clockrate='90000' name='MPA' id='97'/>
   <payload-type clockrate='16000' name='SIREN' id='98'/>
   <payload-type clockrate='8000' name='telephone-event' id='99'/>
  </description>
  </session>
  </iq>

Review: http://reviewboard.digium.com/r/181/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185363 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13985)
Reported by: jcovert
Tested by: phsultan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines

Set the initiator attribute to lowercase in our replies when receiving calls.

This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the 
initiator attribute contains uppercase characters.

(closes issue #13984)
Reported by: jcovert
Patches:
      chan_gtalk.2.patch uploaded by jcovert (license 551)
Tested by: jcovert

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175058 65c4cc65-6c06-0410-ace0-fbb531ad65f3

when a file is invalid from when a file is missing.  This is most important when
we have two configuration files.  Consider the following example:

Old system:
sip.conf     users.conf     Old result               New result
========     ==========     ==========               ==========
Missing      Missing        SIP doesn't load         SIP doesn't load
Missing      OK             SIP doesn't load         SIP doesn't load
Missing      Invalid        SIP doesn't load         SIP doesn't load
OK           Missing        SIP loads                SIP loads
OK           OK             SIP loads                SIP loads
OK           Invalid        SIP loads incompletely   SIP doesn't load
Invalid      Missing        SIP doesn't load         SIP doesn't load
Invalid      OK             SIP doesn't load         SIP doesn't load
Invalid      Invalid        SIP doesn't load         SIP doesn't load

So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed.  Worse yet, the old
system would do this with no indication that anything was even wrong.

(closes issue #10690)
 Reported by: dtyoo
 Patches: 
       20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13310)
Reported by: eliel
Patches:
      chan_gtalk.c.patch uploaded by eliel (license 64)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139281 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #13054)
Reported by: pabelanger
Patches:
      ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130129 65c4cc65-6c06-0410-ace0-fbb531ad65f3

jabber.conf). The actual connection is made when a call comes in
Asterisk.

Apply this fix to Jingle too.

Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.

(closes issue #12085)
Reported by: junky
Tested by: phsultan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119741 65c4cc65-6c06-0410-ace0-fbb531ad65f3

them to the parser ;
- report Gtalk error messages from a buddy to the console.

This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.

Thank you to PH for his great help!

(closes issue #12647)
Reported by: PH
Patches:
      trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118020 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- 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: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines

When modules are embedded, they take on a different name, without the ".so"
extension.  Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114709 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line

Free newly allocated channel before returning
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112821 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines

Prevent call connections when codecs don't match.

(closes issue #10604)
Reported by: keepitcool
Patches:
      branch-1.4-10604-2.diff uploaded by phsultan (license 73)
Tested by: phsultan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112785 65c4cc65-6c06-0410-ace0-fbb531ad65f3