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Yaroslav Panych | 24 May 14:23
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Reload module

Hi

Is there any possibility to load/unload/reload specific module from
other C module?
Or at least how to run cli command?
I urgently need such routines, because my module periodically updates
some configuration files and must make modules to reread that files.
Realtime configuration is not usable in my case.

regards, Yaroslav

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Mark Michelson | 24 May 00:46
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[Code Review] Help mitigate reinvite glares in the SIP channel driver

This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/1946/

Review request for Asterisk Developers.
By Mark Michelson.

Description

There are times where multiple Asterisk servers are peered together over SIP. In such situations, it is possible for both Asterisk servers to attempt to send direct media reinvites to each other simultaneously. This results in a glare situation in which each of the Asterisk servers sends a 491 to the other. After a waiting period, the reinvites are re-attempted. This waiting period can potentially be distracting since it can cause the media to take multiple seconds to finalize, especially if more than 2 Asterisk servers are involved. This patch introduces a new SIP peer option called "directmedia_outgoing". If enabled, then when communicating with the peer, Asterisk will only attempt to send reinvites if the call direction is outgoing. The assumption is that the peer Asterisk server will also have this setting enabled. This way, when the two Asterisk servers communicate, they will never attempt to send direct media reinvites to each other. Instead, it will always be the peer that placed the call that will send the direct media reinvite.

Testing

I have tested this by running two Asterisk servers and ensuring that the option was honored and that the media streams were still set up properly.

Diffs

  • /trunk/channels/chan_sip.c (367417)
  • /trunk/channels/sip/include/sip.h (367417)
  • /trunk/configs/sip.conf.sample (367417)

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Mark Michelson | 23 May 22:46
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[Code Review] Add unique message IDs to IMAP voicemail

This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/1945/

Review request for Asterisk Developers.
By Mark Michelson.

Description

This review has two main parts to it. 1) IMAP voicemail storage now supports unique message IDs like the other storage backends. 2) Old IMAP voicemails that do not have a unique message ID can be updated to have one now. This involves deleting the old message, creating a new one, and then storing that message in the appropriate folder. Comments in the code explain what is going on.

Testing

Admittedly, this has only undergone a compilation test.

Diffs

  • /team/mmichelson/trunk-digiumphones/apps/app_voicemail.c (367360)

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Paul Belanger | 23 May 21:25
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Re: [svn-commits] mmichelson: branch 1.8 r367002 - in /branches/1.8: channels/ include/asterisk...

On 12-05-18 12:54 PM, SVN commits to the Digium repositories wrote:
> Author: mmichelson
> Date: Fri May 18 11:53:47 2012
> New Revision: 367002
>
> URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=367002
> Log:
> Fix memory leak of SSL_CTX structures in TLS core.
>
> SSL_CTX structures were allocated but never freed. This was a bigger
> issue for clients than servers since new SSL_CTX structures could be
> allocated for each connection. Servers, on the other hand, typically
> set up a single SSL_CTX for their lifetime.
>
> This is solved in two ways:
>
> 1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is
> freed so that a new one can take its place.
> 2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
> been added so that servers can properly free their SSL_CTXs.
>
> (issue ASTERISK-19278)
>
You may have to revisit this commit, it break compiling on Ubuntu 10.04, 
see below.

---
tcptls.o: In function `ast_ssl_teardown':
/home/pabelanger/svn/digium/asterisk/testing/1.8/main/tcptls.c:407: 
undefined reference to `SSL_CTX_free'
/home/pabelanger/svn/digium/asterisk/testing/1.8/main/tcptls.c:407: 
undefined reference to `SSL_CTX_free'
collect2: ld returned 1 exit status
make[1]: *** [asterisk] Error 1
make: *** [main] Error 2

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Asterisk Development Team | 23 May 16:44
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Planned service outage for community services

On May 31, 2012 from approximately 9:00AM to 12:00PM (Central Daylight 
Time, GMT-5), the servers that Digium uses to provide many services to 
the Asterisk community will be relocated. This will mean that these 
services will be unavailable during most, if not all, of this time 
window. Once the move is complete, the services will be available again, 
with no user-visible changes.

The services affected include:

bamboo.asterisk.org
code.asterisk.org
downloads.digium.com
downloads.asterisk.org
git.asterisk.org
issues.asterisk.org
packages.asterisk.org
reviewboard.asterisk.org
svn.asterisk.org
svnview.digium.com
wiki.asterisk.org

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Birger Harzenetter | 23 May 15:38
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[Code Review] Remove AST_FLAG_ANSWERED_ELSEWHERE, duplicating the functionality of AST_CAUSE_ANSWERED_ELSEWHERE

This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/1944/

Review request for Asterisk Developers.
By Birger Harzenetter.

Description

While 'real' channels use AST_CAUSE_ANSWERED_ELSEWHERE, local channels and queues use AST_FLAG_ANSWERED_ELSEWHERE for the same purpose. This patch replaces all occurrences of the flag by the cause code, removing the duplicated functionality and the flag itself.

Diffs

    View Diff

    		 
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    Matt Jordan | 22 May 19:22
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    [Code Review] Update a peer's lastmsgssent value appropriately

    This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/1939/

    Review request for Asterisk Developers and irroot.
    By Matt Jordan.

    Description

    In prior versions of Asterisk, the lastmsgssent value was used to track whether or not a peer had received MWI notifications. Since chan_sip polled for the notifications itself, the value was rather important. When MWI notifications were changed to use the event notification framework, the value was no longer useful for anything other then reporting through the CLI or AMI events. Hence, in Asterisk 10 and trunk, the value was completely removed. Unfortunately, in Asterisk 1.8, the value was not removed; instead, it is set to a value of -1 and never updated. Since the lower 16 bits are used for old messages and the upper 16 bits are used for new messages, this results in the following being displayed for 'sip show peer foo': LastMsgsSent : 32767/65535 Normally, I'd suggest that we remove the field from Asterisk 1.8 and call it a day. However, since this field was supplied to users via AMI and the CLI, doing so breaks backwards compatibility. This patch is a modification of a patch originally supplied by irroot on ASTERIS-17866. It re-implements updating of lastmsgssent when an MWI notification is sent to a SIP peer. Note that the original patch had to be modified slightly due to changes in sip_send_mwi_to_peer (and that up to three threads can be involved in accessing lastmsgssent means that, at the very least, the peer really should be locked when we update it). If we decide that this isn't worth it, we should instead remove the field from Asterisk 1.8, as reporting an erroneous value isn't terribly useful.

    Testing

    Tested with two SIP realtime peers. One peer without voicemail continued to display the 32767/65535 value - which is expected, as that value indicates that we haven't sent any message notifications. Another peer, with voicemail, correctly showed the new/old message counts post registration, and displayed the counts correctly after MWI notifications were sent when new voicemails were left or listened to.

    Diffs

    • /branches/1.8/channels/chan_sip.c (367134)

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    Yaroslav Panych | 22 May 16:51
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    Maximum allowed/recommended size of AMI action?

    Hi

Developing my custom AMI action for Asterisk. This action intended to
process relatively big block of data(a few KBs)  received as parameter
in action.  So natural question: how long blocks of data I can send
with action and Asterisk will not suffer any side-effects? Or I should
serialise data transfer using separate action?
Tried to search in source  - found only limitation for number of
headers in action - up to 128. But no total size.

Any ideas?

regards, Yaroslav

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    Alistair Cunningham | 22 May 13:44
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    Urgent development consultancy wanted

    We have a customer running Asterisk 1.8.7.1 who is suffering from stuck 
calls. The scenario is:

1. A call comes in from the PSTN via SIP.
2. We do a Dial() to a local channel.
3. In the local channel, we do a Dial() to a SIP URI which is a phone 
registered to OpenSIPS on a different machine.
4. The phone rings (and perhaps answers).
5. The caller hangs up.
6. Sometimes one of the channels (either the inbound channel or the 
local channel) never gets hung up, the "h" extension never gets called 
for it, and the channel remains in "core show channels" until Asterisk 
is restarted.

We're looking for a developer who is able to debug this urgently, 
preferably today. If anyone is available and has expertise at debugging 
this problem, please email me off-list with details of exactly when 
you're available, and of course your hourly rate.

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    opticron | 21 May 23:10
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    [Code Review] Add tests for the IAX2 implementation of the HANGUPCAUSE hash

    This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/1942/

    Review request for Asterisk Developers.
    By opticron.

    Description

    This tests that the proper hangup information is provided across local channels and when dials are branched.

    Testing

    It's a test. It tests.
    Bugs: SWP-4223

    Diffs

    • asterisk/trunk/tests/iax2/hangupcause/configs/ast1/extensions.conf (PRE-CREATION)
    • asterisk/trunk/tests/iax2/hangupcause/configs/ast1/iax.conf (PRE-CREATION)
    • asterisk/trunk/tests/iax2/hangupcause/configs/ast2/extensions.conf (PRE-CREATION)
    • asterisk/trunk/tests/iax2/hangupcause/configs/ast2/iax.conf (PRE-CREATION)
    • asterisk/trunk/tests/iax2/hangupcause/run-test (PRE-CREATION)
    • asterisk/trunk/tests/iax2/hangupcause/test-config.yaml (PRE-CREATION)
    • asterisk/trunk/tests/iax2/tests.yaml (3229)

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    opticron | 21 May 23:10
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    [Code Review] Add IAX2 support for the new HANGUPCAUSE hash

    This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/1941/

    Review request for Asterisk Developers.
    By opticron.

    Description

    Add the IAX2 implementation of the "Who Hung Up?" work for Asterisk 11. Numeric cause codes are provided for messages in which they're expected. Additionally, methods of generating descriptions of frame types and subclasses have been exposed.

    Testing

    See tests in Review 1942.
    Bugs: SWP-4222

    Diffs

    • trunk/channels/chan_iax2.c (367194)
    • trunk/include/asterisk/frame.h (367194)
    • trunk/main/frame.c (367194)

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    Gmane