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Author SHA1 Message Date
russell
6e96b01e27 Merged revisions 72112 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72112 | russell | 2007-06-27 11:34:24 -0500 (Wed, 27 Jun 2007) | 3 lines

Only output debug information related to RTCP timestamps when RTCP debug
is turned on (issue #10066, patch by me)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72113 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27 16:38:12 +00:00
qwell
cad3826385 Merged revisions 71915 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71915 | qwell | 2007-06-26 15:36:09 -0500 (Tue, 26 Jun 2007) | 4 lines

Don't dereference a pointer that may be NULL here.

Issue 10017.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71916 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-26 20:36:50 +00:00
russell
6116c57c53 Convert so more logging to ast_debug (issue #10045, dimas)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71557 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-25 13:42:51 +00:00
russell
438cb82e0f Conversions to ast_debug()
(issue #9984, patches from eliel and dimas)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71338 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-24 18:51:41 +00:00
file
19ab5f94b5 Behold the magic of casting!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71146 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-22 16:14:00 +00:00
murf
c0200463dc Merged revisions 71063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun 2007) | 1 line

My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone.
This is for bug 10016. (plus a small fix to rtp, to elim a compiler warning (dev mode))
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71093 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-22 15:15:35 +00:00
qwell
176dfa7845 Add manager events for RTCP statistics.
Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
  This permission was discussed on the -dev mailing list some months back.

Issue 8613, patch by johann8384, with some minor changes by me.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70961 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-21 23:07:20 +00:00
file
1034354f34 Merged revisions 70727 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70727 | file | 2007-06-21 11:22:39 -0400 (Thu, 21 Jun 2007) | 2 lines

Do not Packet2Packet bridge if packetization settings do not allow it. (issue #9117 reported by phsultan)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70729 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-21 15:25:13 +00:00
file
c677d2618b Merged revisions 70360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70360 | file | 2007-06-20 13:52:57 -0400 (Wed, 20 Jun 2007) | 2 lines

Put the speex packetization values back in but disable it when setting up the smoother.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70361 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-20 17:55:09 +00:00
file
9728b05eb1 Merged revisions 70003 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70003 | file | 2007-06-19 13:07:40 -0400 (Tue, 19 Jun 2007) | 10 lines

Merged revisions 69992 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 lines

Handle the CC field in the RTP header. (issue #9384 reported by DoodleHu)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70006 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-19 17:09:20 +00:00
file
34747bbb82 Merged revisions 68922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68922 | file | 2007-06-12 10:23:11 -0400 (Tue, 12 Jun 2007) | 10 lines

Merged revisions 68921 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 lines

Bring RTP back to Asterisk at the end of a native bridge no matter what.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68923 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-12 14:26:12 +00:00
russell
ae627acb2f Fix a bunch of doxygen errors and document more things
(issue #9842, snuffy)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68339 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-07 23:07:25 +00:00
tilghman
eb5d461ed4 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06 21:20:11 +00:00
file
1545e6dfd5 Merged revisions 67650 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67650 | file | 2007-06-06 09:30:25 -0400 (Wed, 06 Jun 2007) | 10 lines

Merged revisions 67649 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 lines

Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67651 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06 13:32:11 +00:00
russell
1612102bf5 Merged revisions 67071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67071 | russell | 2007-06-04 16:47:36 -0500 (Mon, 04 Jun 2007) | 2 lines

Add a missing \n.  (pointed out by jcmoore on IRC)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67072 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-04 21:48:15 +00:00
file
254ab3e0b8 Merged revisions 66437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66437 | file | 2007-05-29 12:44:34 -0400 (Tue, 29 May 2007) | 2 lines

Handle cases where a frame may have no data. (issue #9519 reported by dmb)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66438 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-29 16:46:49 +00:00
russell
f97bf5fd4d Make this build on *my* machine again, and hopefully not break others.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65964 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24 18:07:56 +00:00
file
fc3684d573 Merged revisions 65863 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2 lines

I like it when the RTP stack compiles myself...

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65869 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24 15:10:13 +00:00
russell
32977d176e Merged revisions 65842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines

Fix the calculation of the RTT for RTCP.  The previous code would result in
oscillating and incorrect data.  Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65843 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24 14:50:25 +00:00
russell
3d2428efd4 Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62457 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30 16:16:26 +00:00
qwell
f0cc154be0 Merged revisions 61707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines

Avoid invalid seqno cycling detection.

Per comment from Dave Troy:
 This adds back in some simple typecasting I had in an earlier version
 which I realize now may be breaking things.

Issue #9554.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61708 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20 21:37:04 +00:00
russell
59c77fe14b Merged revisions 61697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) | 2 lines

Remove a stray debug message introduced by a recent commit.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61698 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20 20:43:05 +00:00
oej
ba90ed12c2 Merged revisions 61676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2 lines

Clean upp formatting, add some doxygen stuff while we're in cleaning mode... Thanks Kevin!

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61677 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-18 20:48:13 +00:00
oej
fbc046fa8d Merged revisions 61674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2 lines

Issue #9554 - Improve RTCP (Dave Troy)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61675 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-18 20:39:31 +00:00
russell
ab354dee19 Merged revisions 59358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59358 | russell | 2007-03-29 12:17:41 -0500 (Thu, 29 Mar 2007) | 13 lines

Merged revisions 59357 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines

If an error occurs when reading from an RTP socket, and the error code does not
indicate that we should try again, then return NULL instead of a "null frame".
This will prevent Asterisk from trying over and over again, and eventually
causing the system to crash.  (issue #8285, john)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59359 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-29 17:20:43 +00:00
russell
c04c322b06 Merged revisions 59207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines

The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
because they get set in sip_hangup.  So, there are common situations where
the variables will not be available in the dialplan at all.  So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59208 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26 17:51:27 +00:00
file
5bf0e6b6cd Merged revisions 58783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2 lines

Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58784 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-12 01:22:29 +00:00
file
e676dc69e5 Merged revisions 58436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 lines

Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58437 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-08 18:05:54 +00:00
file
853d5a9163 Merged revisions 58240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines

Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58241 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-07 17:55:11 +00:00
file
6ce2df8c8e Merged revisions 57768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 lines

Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@57769 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05 03:24:18 +00:00
oej
10edb20a8e Doxygen additions, corrections
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56665 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24 20:29:41 +00:00
oej
4e2960819a Adding Realtime Text support (T.140) to Asterisk
T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired. 
Feels very much like the old Unix talk application.

This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.

A big thank you to everyone involved in this.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54838 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16 13:35:44 +00:00
file
bec9a4355a Merged revisions 53434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53434 | file | 2007-02-07 12:53:03 -0500 (Wed, 07 Feb 2007) | 2 lines

We can not reliably do P2P bridging with DTMF passing back with compensation if we need to listen for DTMF frames. (issue #8962 reported by caio1982)

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2007-02-07 17:57:37 +00:00
russell
3b679d120a Merged revisions 53429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) | 7 lines

When parsing the NTP timestamp in a sender report message, you are supposed to
take the low 16 bits of the integer part, and the high 16 bits of the
fractional part.  However, the code here was erroneously taking the low 16 bits
of the fractional part.  It then shifted the result 16 bits down, so the result
was always zero.  This fix makes it grab the appropriate high 16 bits, instead.
(issue #8991, pointed out by andre_abrantes)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53431 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-07 17:46:42 +00:00
file
3d3b2c6ec1 Merged revisions 53120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2 lines

Correct a copy/pasted error message line for RTCP.

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2007-02-02 17:16:05 +00:00
file
a77c668f51 Merged revisions 53052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines

When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party.

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2007-02-01 00:24:50 +00:00
file
2311796b27 Merged revisions 53050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2 lines

Add more frame types to forward in the RTP bridge loops.

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2007-02-01 00:23:19 +00:00
russell
7f146a4756 Merged revisions 53040 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53040 | russell | 2007-01-31 11:45:05 -0600 (Wed, 31 Jan 2007) | 11 lines

Merged revisions 53039 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | 3 lines

Use the proper format string to print unsigned values in the rtp debug output.
(issue #8954, wmis)

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2007-01-31 17:45:43 +00:00
russell
ddc640ed5d Merged revisions 52645 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | 6 lines

Fix a problem with packet-to-packet bridging and DTMF mode translation.  P2P
bridging can only be used when the DTMF modes don't match if the core is
monitoring DTMF in both directions.  Then, the core will handle the translation.
Otherwise, this bridging method can not be used.
(issue #8936)

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2007-01-29 21:27:34 +00:00
file
ef80da3edc Only use locking for bridge information if intense P2P bridging is enabled.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@52635 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-29 21:03:07 +00:00
file
363b34c1e9 Change RTP protos list to be read/write. Most of the time it's only going to be read so making it use mutex locks was a waste.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51467 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-22 23:53:16 +00:00
file
98bcb4fcb0 Make the RTP stack better conform to coding guidelines. (issue #8679 reported by johann8384)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51463 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-22 23:51:42 +00:00
russell
f91595d103 Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines

Merge the changes from the /team/group/vldtmf_fixup branch.

The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged.  So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio.  However,
since there was no audio coming in, the DTMF_END was never generated.  This
caused DTMF based features to no longer work.

To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf).  If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.

Channel drivers also now get passed the length of the digit to their digit_end
callback.  This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.

(issue #8597, maybe others...)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19 18:06:03 +00:00
rizzo
3c6e4ff612 in the interest of portability, avoid using %zd when all
we need is to print is an integer that fits in 16 bits.



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2007-01-19 17:48:48 +00:00
file
58d2e51da6 Merged revisions 51211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51211 | file | 2007-01-17 19:18:44 -0500 (Wed, 17 Jan 2007) | 2 lines

Pass data as well for hold/unhold/vidupdate frames. (issue #8840 reported by mdu113)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51212 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-18 00:20:50 +00:00
file
799b67a560 Merged revisions 51182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51182 | file | 2007-01-17 01:36:41 -0500 (Wed, 17 Jan 2007) | 2 lines

Return the correct result when directly writing out a packet so that the core doesn't then decide to handle it the regular way again. (issue #8833 reported by rcourtna)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-17 06:37:47 +00:00
qwell
174a032d2f Merged revisions 51170 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51170 | qwell | 2007-01-16 18:20:56 -0600 (Tue, 16 Jan 2007) | 4 lines

Fix issue with dtmf continuation packets when the dtmf digit is 0...

Issue 8831

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51171 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-17 00:22:20 +00:00
file
56f828fb5c Merged revisions 50466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines

Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@50467 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-11 05:21:03 +00:00
file
fc31d4eec7 Make callback declaration match one used in trunk.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@50056 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-08 20:10:23 +00:00
file
a5f38dd107 Merged revisions 50032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r50032 | file | 2007-01-08 13:21:31 -0500 (Mon, 08 Jan 2007) | 2 lines

Disable the more intense packet2packet bridging until the bugs can be worked out.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@50033 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-08 18:23:39 +00:00