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r125333 | kpfleming | 2008-06-26 10:50:07 -0500 (Thu, 26 Jun 2008) | 13 lines
Merged revisions 125327 via svnmerge from
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r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines
ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number
(issue AST-80)
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r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) | 18 lines
Merged revisions 125132 via svnmerge from
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
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r124316 | tilghman | 2008-06-20 15:17:04 -0500 (Fri, 20 Jun 2008) | 16 lines
Merged revisions 124315 via svnmerge from
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r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008) | 8 lines
When using a Local channel, started by a call file, with a destination of an
AGI script, the AGI script does not always get notified of a hangup if the
underlying channel hangs up early.
(closes issue #11833)
Reported by: IgorG
Patches:
local_hangup-v1.diff uploaded by IgorG (license 20)
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r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17 Jun 2008) | 5 lines
Updates all usages of ast_tcptls_session_instance to be managed by reference counts so that they only get destroyed when all threads are done using
them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
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r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun 2008) | 19 lines
Merged revisions 123333 via svnmerge from
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r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines
Cisco BTS sends SIP responses with a tab between the Cseq number and
SIP request method in the Cseq: header. Asterisk did not handle this
properly, but with this patch, all is well.
(closes issue #12834)
Reported by: tobias_e
Patches:
12834.patch uploaded by putnopvut (license 60)
Tested by: tobias_e
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r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008) | 16 lines
Merged revisions 123110 via svnmerge from
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r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines
People expect that if "hasvoicemail" is set in users.conf, even if "mailbox"
isn't set, that SIP will detect a mailbox.
(closes issue #12855)
Reported by: PLL
Patches:
20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: PLL
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r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) | 14 lines
Merged revisions 122919 via svnmerge from
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r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines
Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP.
(closes issue #12803)
Reported by: lanzaandrea
Patches:
chan_sip.c.diff uploaded by lanzaandrea (license 496)
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r121230 | mmichelson | 2008-06-09 10:08:58 -0500 (Mon, 09 Jun 2008) | 27 lines
Merged revisions 121229 via svnmerge from
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(Note that this is being merged to trunk/1.6.0 because
it may affect non-callback agents with ackcall set)
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r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines
A unique situation of timeouts brought forth a failure situation for
autologoff in chan_agent. If using AgentCallbackLogin-style agents,
then if the timeout specified by the Dial() to reach the agent's phone
was shorter than the timeout specified in queues.conf, then autologoff
would only work if the caller hung up while the agent's phone was ringing.
This patch allows autologoff to work in this situation when the call in
queue transfers to the next available agent (as it would have if the timeout
in queues.conf were less than the timeout in the Dial()).
(closes issue #12754)
Reported by: Rodrigo
Patches:
12754.patch uploaded by putnopvut (license 60)
Tested by: Rodrigo
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r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) | 4 lines
This was accidentally reverted.
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
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r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) | 16 lines
Merged revisions 120863,120885 via svnmerge from
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r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
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r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
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r119839 | russell | 2008-06-02 15:08:24 -0500 (Mon, 02 Jun 2008) | 15 lines
Merged revisions 119838 via svnmerge from
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r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines
Revert a change made for issue #12479. This change caused a regression such that
a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore.
(closes issue #12770)
Reported by: dagmoller
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r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines
Do not link the guest account with any configured XMPP client (in
jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
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r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008) | 9 lines
Merged revisions 119585 via svnmerge from
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r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line
Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44).
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r119637 | crichter | 2008-06-02 04:35:04 -0500 (Mon, 02 Jun 2008) | 9 lines
Merged revisions 119636 via svnmerge from
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r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line
fixed compile issue when dev-mode is enabled
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r119239 | russell | 2008-05-30 07:59:11 -0500 (Fri, 30 May 2008) | 23 lines
Merged revisions 119238 via svnmerge from
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r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines
Merged revisions 119237 via svnmerge from
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r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines
- Instead of only enforcing destination call number checking on an ACK, check
all full frames except for PING and LAGRQ, which may be sent by older versions
too quickly to contain the destination call number.
(As suggested by Tim Panton on the asterisk-dev list)
- Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ
from being sent before the destination call number is known.
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r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008) | 11 lines
Merged revisions 118953 via svnmerge from
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r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines
Add some debugging code that ensures that when we do deadlock avoidance, we
don't lose the information about how a lock was originally acquired.
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r118957 | tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10 lines
Merged revisions 118954 via svnmerge from
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r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008) | 2 lines
Define also when not DEBUG_THREADS
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r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines
Merged revisions 118646 via svnmerge from
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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann
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r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008) | 20 lines
Merged revisions 118251 via svnmerge from
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r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines
Realtime flag affects construction in multiple ways, so consulting whether
rtcachefriends was set was done too soon (needed to be done inside build_peer,
not just as a flag to build_peer).
Also, fullcontact needed to be reconstructed, because realtime separates the
embedded ';' into multiple fields.
(closes issue #12722)
Reported by: barthpbx
Patches:
20080525__bug12722.diff.txt uploaded by Corydon76 (license 14)
Tested by: barthpbx
(Much of the discussion happened on #asterisk-dev for diagnosing this issue)
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