There was already code for other signaling types in dahdi_handle_event to
handle dialing if a dial operation dial string was present. Simply add
SIG_EMWINK to the list.
(closes issue #14434)
Reported by: araasch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new dahdi_wait function to specifically wait for the wink
event. If the wink is not eventually received the channel is hung up.
(closes issue #14434)
Reported by: araasch
Patches:
emwinkmod uploaded by araasch (license 693)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Copied from Review board:
In issue 12434, the reporter describes a situation in which audio and video
is offered on the call, but because videosupport is disabled in sip.conf,
Asterisk gives no response at all to the video offer. According to RFC 3264,
all media offers should have a corresponding answer. For offers we do not
intend to actually reply to with meaningful values, we should still reply
with the port for the media stream set to 0.
In this patch, we take note of what types of media have been offered and
save the information on the sip_pvt. The SDP in the response will take into
account whether media was offered. If we are not otherwise going to answer
a media offer, we will insert an appropriate m= line with the port set to 0.
It is important to note that this patch is pretty much a bandage being
applied to a broken bone. The patch *only* helps for situations where video
is offered but videosupport is disabled and when udptl_pt is disabled but
T.38 is offered. Asterisk is not guaranteed to respond to every media offer.
Notable cases are when multiple streams of the same type are offered.
The 2 media stream limit is still present with this patch, too.
In trunk and the 1.6.X branches, things will be a bit different since Asterisk
also supports text in SDPs as well.
(closes issue #12434)
Reported by: mnnojd
Review: https://reviewboard.asterisk.org/r/311
Review: https://reviewboard.asterisk.org/r/313
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously overlap dialing could only be turned on or off for both incoming and
outgoing calls. New parameters incoming, outgoing, and both have been added to
allow further control. There is no change in default behavior with these new
options and allows in band DTMF to be accepted in one direction if required.
(closes issue #14471)
Reported by: eboscani
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some ITSP make use of the "Anonymous" display name to detect a
requirement to withhold caller id across the PSTN. This does
not work if the display name is "Unknown".
(closes issue #14465)
Reported by: Nick_Lewis
Patches:
chan_sip.c-callerpres.patch uploaded by Nick (license 657)
chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
..........
Fixed chan_misdn crash because mISDNuser library is not thread safe.
With Asterisk the mISDNuser library is driven by two threads concurrently:
1. channels/misdn/isdn_lib.c::manager_event_handler()
2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher()
Calls into the library are done concurrently and recursively from
isdn_lib.c.
Both threads can fiddle with the master/child layer3_proc_t lists. One
thread may traverse the list when the other interrupts it and then removes
the list element which the first thread was currently handling. This is
exactly what caused the crash. About 60 calls were needed to a Gigaset
CX475 before it occurred once.
This patch adds locking when calling into the mISDNuser library.
This also fixes some cb_log calls with wrong port parameter.
JIRA ABE-1913
Patches: misdn-locking.patch (Modified with mostly cosmetic changes)
..........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* issue #14355 - Crash if attempt to transfer a call to an application.
Masquerade the other pair of the four asterisk channels involved in the
two calls. The held call already must be a bridged call (not an
applicaton) or it would have been rejected.
* issue #14692 - Held calls are not automatically cleared after transfer.
Allow the core to initate disconnect of held calls to the ISDN port. This
also fixes a similar case where the party on hold hangs up before being
transferred or taken off hold.
* JIRA ABE-1903 - Orphaned held calls left in music-on-hold.
Do not simply block passing the hangup event on held calls to asterisk
core.
* Fixed to allow held calls to be transferred to ringing calls.
Previously, held calls could only be transferred to connected calls.
* Eliminated unused call states to simplify hangup code.
* Eliminated most uses of "holded" because it is not a word.
(closes issue #14355)
(closes issue #14692)
Reported by: sodom
Patches:
misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines
Ensure apathetic replies are sent out on the proper socket.
chan_iax2 supports multiple address bindings. The send_apathetic_reply()
function did not attempt to send its response on the same socket that the
incoming message came in on.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From the review board request:
The fix from review 298 has exposed a new bug in chan_sip.
When we hang up an outgoing call, we first will dump all the outstanding
packets on the sip_pvt using __sip_pretend_ack. Then, if we can, we send
a CANCEL. The problem with this is that since destroyed all the outstanding
packets on the dialog, we cannot match the incoming 487 response to our
INVITE. Because we cannot match the response, we do not send an ACK.
To correct this, instead of using __sip_pretend_ack, I have changed the code
to loop through the list of packets and call __sip_semi_ack on each one
instead. This causes us to stop retransmitting the requests, but we still have
them around in case we get responses for them.
When discussing this earlier today with Josh Colp, we both agreed that in the
majority of cases, this would be enough of a fix. However, we also agreed that
we should have a safety net in place in case we never receive a response to
our initial INVITE. To handle this, I have modified __sip_autodestruct to
behave similar to the way it does in Asterisk 1.4. If there are outstanding
packets on the sip_pvt, the needdestroy flag is not set, and the last request
on the dialog was either a CANCEL or BYE, then we set the needdestroy flag and
reschedule destruction for 10 seconds in the future. If, though, the
needdestroy flag is set, then we use __sip_pretend_ack to kill the remaining
outstanding packets so that the monitor thread can destroy the sip_pvt.
I ran two separate tests. First, I placed a call from my Aastra phone to my
Polycom phone. I hung up the Aastra before the Polycom answered. I verified
through sip debug output that Asterisk properly ACKed the 487 received from the
Polycom.
For my second test, I set up a SIPp UAS scenario so that it would send a 200 OK
in response to a CANCEL but would not send a 487 for the original INVITE. I
verified that after about 40 seconds, Asterisk properly cleans up the outgoing
sip_pvt for the call.
Review: https://reviewboard.asterisk.org/r/308
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an endpoint sends two registration requests in a very short
period of time with the same nonce, both receive 401 responses
from Asterisk, each with a different nonce (the second 401
containing the current nonce and the first one being stale).
If the endpoint responds to the first 401, it does not match
the current nonce so Asterisk sends a third 401 with a newly
generated nonce (which updates the current nonce)... Now if
the endpoint responds to the second 401, it does not match the
current nonce either and Asterisk sends a fourth 401 with a
newly generated nonce... This loop goes on and on.
There appears to be a simple fix for this. If the nonce from
the request does not match our nonce, but is a good response
to a previous nonce, instead of sending a 401 with a newly
generated nonce, use the current one instead. This breaks
the loop as the nonce is not updated until a response is
received. Additional logic has been added to make sure no
nonce can be responded to twice though.
(closes issue #15102)
Reported by: Jamuel
Patches:
patch-bug_0015102 uploaded by Jamuel (license 809)
nonce_sip.diff uploaded by dvossel (license 671)
Tested by: Jamuel
Review: https://reviewboard.asterisk.org/r/289/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this change, we make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will have the proper
Route headers in them.
(closes issue #14725)
Reported by: ibc
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
(i.e. When libpri generates the event PRI_EVENT_ANSWER.)
(closes issue #15420)
Reported by: scottbmilne
Patches:
bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
Tested by: scottbmilne, alecdavis
(closes issue #15416)
Reported by: avinoash
(closes issue #15389)
Reported by: alecdavis
This patch should also fix the following issue:
(issue #15205)
Reported by: vinsik
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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
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
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
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
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
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
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
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
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If we already have an address for a peer, and we are reloading the sip
configuration, try to use that address to contact the peer, instead of
getting it from the Contact.
(closes issue #15194)
Reported by: ibc
Patches:
sip.patch uploaded by eliel (license 64)
Tested by: manwe
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated
(or it passes through unauthenticated) the proper nat flag is set.
(closes issue #13823)
Reported by: dimas
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement.
(closes issue #15032)
Reported by: guillecabeza
Patches:
chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380)
Tested by: guillecabeza
(closes issue #14216)
Reported by: Andrey Sofronov
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
