mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
13657 Commits (3913dcbc40039ca3c1942b7300fc45daeb4970ea)
| Author | SHA1 | Message | Date |
|---|---|---|---|
David Vossel
|
259998a286 |
Changing ast_samp2tv to not use floating point.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205599 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
beaf6217b3 |
Fixes 8khz assumptions
Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
2e330f772c |
moving ast_devstate_to_extenstate to pbx.c from devicestate.c
ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205409 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
16d3415cf3 |
Prevent phantom calls to queue members.
If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jason Parker
|
c43a1be4ec |
Update config.guess and config.sub from the savannah.gnu.org git repo.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205288 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
9f4c452028 |
ast_samp2tv needs floating point for 16khz audio
In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205215 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
0b1f3adf7f |
Add redirection warnings for the invalid language codes previously removed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
33f54353ba |
Make OpenSSL usage thread-safe.
OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205149 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Richard Mudgett
|
202f9967c6 |
Removed confusing warning message "Got Busy in Connected State"
If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
David Vossel
|
bdada0dce1 |
moving device state functions from pbx.h to devicestate.h to sync with other branches
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204755 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
David Vossel
|
4c99b19973 |
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204681 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
e8f0570118 |
More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar.
(closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204556 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Jason Parker
|
b3e413e910 |
Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204474 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
60270012a9 |
"tw" is the language specification for Twi (from Ghana) not Taiwanese.
(closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
e5bef05d8f |
Add error message so that it is clear why a SIP peer was not processed when
a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204300 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
439ce618c5 |
Fix build oops.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204246 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
9589d9fb2e |
Fix a problem where chan_sip would ignore "old" but valid responses.
chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204243 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
399bd49b7d |
Revision 189537 was supposed to make 1.4 more correct. Instead, it broke func_odbc. Reverting.
(closes issue #15317, issue #14614) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204170 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
David Vossel
|
4f3580b882 |
segfault after SPINLOCK schedule delete
Using the SPINLOCK schedule delete macro can result in the iax_pvt lock being given up. This makes it possible for the iax_pvt to dissappear when we thought we held the mutex the entire time. To resolve this, the iax_pvt's ref count is incremented. (closes issue #15377) Reported by: aragon Patches: iax_spin_issue_1.4.diff uploaded by dvossel (license 671) Tested by: aragon, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204067 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
a3848ec74c |
Place unlock of mutex in an else block so that it does not get unlocked twice.
(closes issue #15400) Reported by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Richard Mudgett
|
f65dccafb6 |
The ISDN CPE side should not exclusively pick B channels normally.
Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jeff Peeler
|
fc73897bbd |
Make sure to recreate the dahdi pseudo channel after dahdi restart
(closes issue #14477) Reported by: timking git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203848 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
fe7923abfc |
Don't fast forward past the end of a message.
This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
David Brooks
|
64e75ecf80 |
Fixing voicemail's error in checking max silence vs min message length
Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented as seconds. Also, the inequality was reversed. The warning, if triggered, was "Max silence should be less than minmessage or you may get empty messages", which should have been logged if max silence was greater than minmessage, but the check was for less than. Also, conforming if statement to coding guidelines. closes issue #15331) Reported by: markd Review: https://reviewboard.asterisk.org/r/293/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203719 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Terry Wilson
|
5e3e234df6 |
I didn't see that Mark already fixed the underlying issue!
Yay for removing useless code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
090c2f1e4e |
Fix a case where CDR answer time could be before the start time involving parking.
(closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Terry Wilson
|
03d3fb7a7a |
Don't try to free NULL
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203311 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
140d84dfc5 |
Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
c05d6ceccd |
Resolve a crash related to a T.38 reinvite race condition.
This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Richard Mudgett
|
1ac27cf7ec |
Improved chan_dahdi.conf pritimer error checking.
Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
a1fa4f0391 |
Use the handy UNLINK macro instead of hand-coding the same thing in-line.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202966 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
David Vossel
|
d6106936cb |
MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport
(closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
f76b499923 |
Fix more memory leaks that may result if rtp is not successfully allocated.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202601 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
b0c0c17764 |
Fix potential memory leak in chan_sip when video rtp is not allocated properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202572 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
72b285ed96 |
Report CallerID change during a masquerade.
Reported by: markster git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202496 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
dcfd8d7c7c |
Make Polycom subscription type override check more explicit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
26ba38b8f4 |
Remove an extra debug line left from previous commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202342 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
d31f78a172 |
Fix a situation in which Asterisk would not stop retransmitting 487s.
If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202341 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
1f7d3e9a01 |
Fix a possible infinite loop in SDP parsing during glare situation.
There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202336 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Sean Bright
|
f543251260 |
Since we don't have sip_pvt_lock() in 1.4, we need to use ast_mutex_* directly.
(closes issue #15366) Reported by: loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202153 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Matthew Nicholson
|
e735cdc36b |
Added deadlock protection to try_suggested_sip_codec in chan_sip.c.
Review: https://reviewboard.asterisk.org/r/287/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
David Vossel
|
f17d5d22d2 |
timestamp was being converted to host order as a short rather than a long
(closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
5757b115b3 |
If the "h" extension fails, give it another chance in main/pbx.c.
If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201828 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Russell Bryant
|
fce4a98f7c |
Fix memory corruption and leakage related reloads of non files mode MoH classes.
For Music on Hold classes that are not files mode, meaning that we are executing
an application that will feed us audio data, we use a thread to monitor the
external application and read audio from it. This thread also makes use of the
MoH class object. In the MoH class destructor, we used pthread_cancel() to ask
the thread to exit. Unfortunately, the code did not wait to ensure that the
thread actually went away. What needed to be done is a pthread_join() to ensure
that the thread fully cleans up before we proceed. By adding this one line, we
resolve two significant problems:
1) Since the thread was never joined, it never fully goes away. So, on every
reload of non-files mode MoH, an unused thread was sticking around.
2) There was a race condition here where the application monitoring thread
could still try to access the MoH class, even though the thread executing
the MoH reload has already destroyed it.
(issue #15109)
Reported by: jvandal
(issue #15123)
Reported by: axisinternet
(issue #15195)
Reported by: amorsen
(issue AST-208)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
18 years ago |
|
Mark Michelson
|
03909de702 |
Change the datastore traversal in ast_do_masquerade to use a safe list traversal.
It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
David Vossel
|
86c204f34c |
StopMixMonitor race condition (not giving up file immediately)
StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201423 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
David Brooks
|
ebe2c1829b |
Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read()
Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Kevin P. Fleming
|
b8417b571b |
Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.
When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
|
94fa4d11b5 |
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Eliel C. Sardanons
|
a42ff13c97 |
Show the interface name on error, if it is not found.
If the smdiport specified is not found, show the interface name instead of '(null)'. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |