the existing channel name without the "Zap/" or "DAHDI/" prefix, since our
test string is also without that prefix.
(closes issue #13027)
Reported by: dferrer
Patches:
chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525)
(Slightly modified by me, to compensate for both names)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@137847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been
reported against chan_h323 as well. It seems that the best
solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL.
This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.
(closes issue #13247)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@136062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
was perverted. This change reverts IAX2 to the original meaning, which was,
that the callerid set on the client should be overridden on the server, even if
that means the resulting callerid is blank. In other words, if you set
"callerid=" in the IAX config, then the callerid should be overridden to blank,
even if set on the client. Note that there's a distinction, even on realtime,
between the field not existing (NULL in databases) and the field existing, but
set to blank (override callerid to blank).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@135747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the
static qualifier.
Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@133169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
variable to the block where it is used. This allows one
less #ifdef HAVE_PRI to clutter things up.
Thanks to Tzafrir for pointing this out on #asterisk-dev
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@133038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.
This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.
(closes issue #7403)
Reported by: stephen_dredge
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@132790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
correct registration of AMI actions in chan_dahdi; in zap-only mode, only register the Zap flavors of the actions (and use Zap prefixes for headers and acks), but in dahdi+zap mode, register both Zap and DAHDI flavors of actions
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@132787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@132645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
registration). Related to revisions 132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently tested 1.4 with a backport from those revisions,
so I didn't see this problem (oops).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@132506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This should fix talk call progress on analog lines.
(closes issue #12178)
Reported by: michael-fig
Patches:
20080717__bug12178.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@131790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
just when a remote callerid is set. Also, if not set in the user, allow the
remote CallerID to pass through.
(closes issue #12875)
Reported by: dimas
Patches:
20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@130889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sean Bright
Jeff Peeler
Tilghman Lesher
Kevin P. Fleming
Richard Mudgett
Mark Michelson
Michiel van Baak
Russell Bryant
Jason Parker
Christian Richter
Olle Johansson
Brett Bryant