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r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines
Fixes regression caused by r343635
There was a missing unlock for a function return that is only
present in Asterisk 10 and Asterisk Trunk.
(closes issue ASTERISK-18839)
Reported by: Michael L. Young
Patches:
asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young
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* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
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The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command
pulled the flags from the incorrect global flags page. These are now read from
sip global flags page 0.
(closes issue AST-711)
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ASTERISK-18669
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Changing an object value used as a container key requires removing the
object from the container and reinserting it.
* Created change_callid_pvt() to call instead of build_callid_pvt(). The
change_callid_pvt() will correctly change the dialog callid so the ao2
conainter can explicitly unlink it.
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Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks.
This function now requires that both the peer and associated pvt be unlocked
before it is called for cases where peer and peer->mwipvt form a circular
reference.
(closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/
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A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
because of a deadlock between the dialogs container lock and the RWLOCK of
the events subscription list.
* Create dialogs_to_destroy container to hold dialogs that will be
destroyed.
* Ensure that the event subscription callback will never happen with an
invalid peer pointer by making the event callback removal the first thing
in the peer destructor callback.
NOTE: This particular deadlock will not happen with Asterisk 10, but some
of the changes still apply.
(closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Review: https://reviewboard.asterisk.org/r/1564/
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The original REGISTERTRYING flag, in addition to being impossible to
check, also encroached on the space for the flag above it. This
patch moves the flags that were below REGISTERTRYING back to where
they were as though we had just removed the REGISTERTRYING option.
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This option is not only useless, but has been broken since inception since
the flag was never copied from the peer where it is set to the pvt where
it was checked. RFC 3261 specificially states that you should not send a
provisional response to a non-INVITE request, and if we did fix the code
so that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This patch
removes registertrying option and any code that would have sent a 100
response to a register.
Review: https://reviewboard.asterisk.org/r/1562/
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Changeset r342927 introduced a warning which was only supposed to be
emitted when a found realtime peer had an empty (or no) name. It turned
out that there were some inconsistencies left. Now found peers with an
empty name are explicitly ignored like before r342927 but better.
Reviewed by: Stefan Schmidts, Terry Wilson
Review: https://reviewboard.asterisk.org/r/1560
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There were several problems with the dynamic realtime peer/user lookup
code. The lookup logic had become rather hard to read due to lots of
incremental changes to the realtime_peer function. And, during the
addition of the sipregs functionality, several possibilities for memory
leaks had been introduced. The insecure=port matching has always been
broken for anyone using the sipregs family. And, related, the broken
implementation forced those using sipregs to *still* have an ipaddr
column on their sippeers table.
Thanks Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which caused
the realtime_peer to have a completely unused code path.
This changeset fixes the leaks, the lookup inconsistenties and that
you won't need an ipaddr column on your sippeers table anymore (when
you're using sipregs). Beware that when you're using sipregs, peers
with insecure=port will now start matching!
(closes issue ASTERISK-17792)
(closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes
Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1395
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When an extension is removed from a context, its entry in the pattern match
tree is not deleted. Instead, the extension is marked as deleted. When an
extension is removed and re-added, if that extension is also a prefix of
another extension, several log messages would report an error and did not
check whether or not the extension was deleted before accessing the memory.
Additionally, if the extension was already in the tree but previously
deleted, and the pattern was at the end of a match, the findonly flag was
not honored and the extension would be erroneously undeleted.
Additionaly, it was discovered that an IAX2 peer could be unregistered
via the CLI, while at the same time it could be scheduled for unregistration
by Asterisk. The unregistration method now checks to see if the peer
was already unregistered before continuing with an unregistration.
(closes issue ASTERISK-18135)
Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1526
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The "No D-channels available! Using Primary channel as D-channel anyway!"
WARNING message has been confusing on non-NFAS setups. The message refers
to things that are NFAS specific.
* Changed the warning to several different warnings to be more accurate
for the situation and less confusing as a result:
"No D-channels up! Switching selected D-channel from X to Y.",
"No D-channels up!", and
"D-channel is down!".
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This behavior matches up more closely with the way invite/register/etc are handled.
This patch also modifies some adjacent code for code style compliance. Pretty minor.
(closes issue ASTERISK-17616)
Reported by: Jeremy Kister
Patches:
chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232)
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Google has recently make some changes (again) to their protocol. Rather then
patching asterisk to flip between the two different methods, we now allow both.
Lets hope this keeps Google Voice happy for a while.
(closes issue ASTERISK-18714)
Reported by: Iordan Iordanov
Patches:
chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311)
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Just create an normal API function in strings.h that does the same thing
just to be safe.
ASTERISK-17146
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a Contact URI from a UAS
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If a SIP dial string contains a numeric hostname that is not a peer name,
don't try to resolve it as it is unlikely that someone really means
Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that
create_addr returns -1 if an address isn't resolved so that we don't
attempt to send SIP requests to an address that doesn't resolve.
(closes issue ASTERISK-17146, ASTERISK-17716)
Review: https://reviewboard.asterisk.org/r/1532/
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* Fix potential deadlocks in SIP and IAX blind transfer to parking.
* Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the
parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
parameter). Created ast_park_call_exten() and ast_masq_park_call_exten()
to maintian API compatibility.
* Made masq_park_call() handle a failed ast_channel_masquerade() setup.
* Reduced excessive struct parkeduser.peername[] size.
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If parse_uri was called with an empty URI, some pointers would be
modified and an invalid read could result. This patch avoids calling
parse_uri with an empty contact uri when parsing REGISTER requests.
AST-2011-012
(closes issue ASTERISK-18668)
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(closes issue ASTERISK-18696)
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If a simple switch was started on a device and then a specific call
made (such as redial or speed dial), on timeout of the simple switch
the call would be attempted again. This patch only allows the simple
switch to make a call if the substate is still in the collecting
digits mode.
Also added small debug message to dialAndAactivate sub.
Tested by snuff and myself.
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r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines
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r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines
Quiet RTCP Receiver Reports during fax transmission
RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
code was added to support the bug fix.
(closes issue ASTERISK-18400)
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r340718 | schmidts | 2011-10-13 06:59:50 +0000 (Thu, 13 Oct 2011) | 9 lines
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r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) | 3 lines
storing the route-set also on a 181 response not only on 180,182 or 183.
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Avoid possible jump based on unitialized value
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r340577 | schmidts | 2011-10-12 20:33:37 +0000 (Mit, 12 Okt 2011) | 9 lines
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r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) | 3 lines
Store route-set from provisional SIP responses so early-dialog requests can be routed properly
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r340578 | twilson | 2011-10-12 13:57:19 -0700 (Wed, 12 Oct 2011) | 16 lines
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r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines
Update SIP realtime fullcontact regardless of caching
We should update the fullcontact field in the realtime table whether or
not rtcachefriends is set. There is no reason to treat a non-cached
realtime entity differently than a cached in this regard.
(closes issue ASTERISK-18446)
Reported by: wdoekes
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The PRI channel alarms were initialized with an inverted sense.
(closes issue ASTERISK-18710)
Reported by: Tzafrir Cohen
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(closes issue ASTERISK-18612)
Reported by: Tim Osman
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* Added a CLI "ss7 show channels" command that might prove useful for
future debugging.
* Made the incoming SS7 channel event check and gripe message uniform.
* Made sure that the DNID string for an incoming call is always
initialized.
(issue ASTERISK-17966)
Reported by: Kenneth Van Velthoven
Patches:
jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett
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* Fixed deadlock potential calling dialog_unlink_all() in
__sip_autodestruct(). Found by helgrind.
* Fixed deadlock potential in handle_request_invite() after calling
sip_new(). Found by helgrind.
* The sip_new() function now returns with the created channel already
locked.
* Removed the dead code that starts a PBX in in sip_new(). No sip_new()
callers caused that code to be executed and it was a bad thing to do
anyway.
* Removed unused parameters and return value from dialog_unlink_all().
* Made dialog_unlink_all() and __sip_autodestruct() safely obtain the
owner and private channel locks without a deadlock avoidance loop.
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r340165 | mjordan | 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) | 20 lines
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r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) | 13 lines
Updated chan_sip to place calls on hold if SDP address in INVITE is ANY
This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::.
In this case, the call should be placed on hold. Previously, we checked for
the address being null; this patch keeps that behavior but also checks for
the ANY IP addresses.
Review: https://reviewboard.asterisk.org/r/1504/
(closes issue ASTERISK-18086)
Reported by: James Bottomley
Tested by: Matt Jordan
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Added some data to skinny packet structures to make compatible
with v17. Added protocolversion to device, set on registration
based on the version provided by device.
v17 includes some increased ip space for ip6. This patch increases
ip space in the packets but still only uses ip4. Some packet
structures duplicated (ip4 and ip6 types). ip4 type used unless
version is greater or equal to 17.
Tested by snuff and myself on 7961 with recent 8.5 firmware. Also
tested compatible with old 7960 and older 30VIPs.
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Increase SKINNY_MAX_PACKET to 2000 bytes to handle some messages
in v17 that are greater than the old 1000 bytes. Also add some
useful logging regarding packet and session handling.
A device (with protocol v17) was sending a packet with length
greater than 1000 which resulted in the TCP session being
destroyed and registration being retryed.
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r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) | 8 lines
Return -1 to skinny_session if register rejected.
If device registration is rejected, return -1 so that the session is
destroyed immediately. Previously, a segfault would occur on a
graceful shutdown if a register is rejected and the skinny_session
has not yet timed out.
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r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) | 9 lines
Remove log message on traverse session list.
On destroying a session, a list of sessions is traversed to find the
matching session. For each session not matching, skinny erroneously
logged that the session was not matched. While technically correct
the message was misleading, and tended to indicate errors that
were not there.
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r339942 | igorg | 2011-10-09 08:18:02 +0700 (Вск, 09 Окт 2011) | 12 lines
Merged revisions 339938 via svnmerge from
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r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | 6 lines
Fix compilation issue, caused by missed session structure
(closes issue ASTERISK-18694)
Reported by: alex70
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r339885 | igorg | 2011-10-08 22:46:27 +0700 (Сбт, 08 Окт 2011) | 13 lines
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r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines
Fix segfault in Unistim channel
(closes issue ASTERISK-18638)
Reported by: jonnt
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r339831 | igorg | 2011-10-08 22:01:35 +0700 (Сбт, 08 Окт 2011) | 14 lines
Merged revisions 339830 via svnmerge from
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r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines
Fix char array cast as short array in send_client() function (for ARM
platform)
(closes issue ASTERISK-17314)
Reported by: jjoshua
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There was an issue that the cap and confcap pointers for each line and device
were being memcpy'd so they all pointed to the same ast_format_cap. On
destroying, a segfault occured on the second call to the same struct.
skinny reload now works again as well.
Tested by snuff and myself.
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r339626 | rmudgett | 2011-10-06 12:53:00 -0500 (Thu, 06 Oct 2011) | 25 lines
Merged revisions 339625 via svnmerge from
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r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines
Fix debugging messages generated by 'udptl debug'.
* Makes chan_sip set the tag to the channel name.
* Fixes received debug message sequence number.
* Removed tx/rx debug message type since it was hard coded to 0.
* Made udptl.c logged message header consistent if possible: "UDPTL (%s): ".
* Removed unused rx_expected_seq_no from struct ast_udptl.
(closes issue ASTERISK-18401)
Reported by: Kevin P. Fleming
Patches:
jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Matthew Nicholson
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r339148 | lmadsen | 2011-10-03 15:13:16 -0500 (Mon, 03 Oct 2011) | 14 lines
Merged revisions 339147 via svnmerge from
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r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) | 6 lines
Remove duplicated Maxforwards line in AMI output.
(Closes issue ASTERISK-18637)
Reported by: Jacek Konieczny
Patches:
asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny
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r339088 | twilson | 2011-10-03 11:44:27 -0700 (Mon, 03 Oct 2011) | 17 lines
Merged revisions 339086 via svnmerge from
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r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) | 10 lines
Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places
After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame
is sent when a re-invite happens. If we receive a re-invite from a device
the waitstream_core was not aware of the new control frame and would drop
the call.
(closes issue ASTERISK-18610)
Reported by: Kristijan_Vrban
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r338801 | rmudgett | 2011-09-30 17:06:48 -0500 (Fri, 30 Sep 2011) | 19 lines
Merged revisions 338800 via svnmerge from
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r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) | 12 lines
Fix segfault in analog_ss_thread() not checking ast_read() for NULL.
NOTE: The problem was reported against v1.6.2. It is unlikely to ever
happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely
to be used. The version in sig_analog.c has largely replaced it.
(closes issue ASTERISK-18648)
Reported by: Stephan Bosch
Patches:
jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Stephan Bosch
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--Denna och nedanstående rader kommer inte med i loggmeddelandet--
M channels/chan_sip.c
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ASTERISK-17486 exposed the problem for AMI parsers.
(closes issue ASTERISK-18649)
Reported by: Jacek Konieczny
Patches:
asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny
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Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r338417 | irroot | 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) | 19 lines
Merged revisions 338416 via svnmerge from
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r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines
The rtptimeout setting is ignored on a per peer basis.
Not only is the rtptimeout ignored in some cases but
rtpkeepalive and rtpholdtimeout is affected.
this commit also removes rtptimeout/rtpholdtimeout on
text rtp.
(closes issue ASTERISK-18559)
Review: https://reviewboard.asterisk.org/r/1452
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r338323 | rmudgett | 2011-09-28 17:36:57 -0500 (Wed, 28 Sep 2011) | 12 lines
Merged revisions 338322 via svnmerge from
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r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) | 5 lines
Make duplicate call ptr warning message more helpful.
* Adds the value of the call ptr to the duplicate call ptr message to help
trace why there is a duplicate call ptr.
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r338228 | qwell | 2011-09-28 15:54:35 -0500 (Wed, 28 Sep 2011) | 9 lines
Merged revisions 338227 via svnmerge from
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r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) | 1 line
Add support levels to non-module sections of menuselect (cflags, utils, etc).
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r338225 | rmudgett | 2011-09-28 15:26:39 -0500 (Wed, 28 Sep 2011) | 12 lines
Merged revisions 338224 via svnmerge from
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r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) | 5 lines
Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present.
(closes issue ASTERISK-18357)
Reported by: Matthew Nicholson
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