Re: [asterisk-dev] Asterisk 18.0.0 Now Available

[Joshua C. Colp]

On Tue, Oct 27, 2020 at 6:14 PM John Kiniston 
wrote:

> Is anyone else seeing menuconfig give the wrong description
> app_audiosocket and chan_audiosocket selections with this release?
>
> I've tried on two systems and I'm seeing the same thing, If I highlight
> app_audioosocket I get a description of  AST_MODULE_INFO(
> and chan_audiosocket has a description of
> AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER,
>
>
>
> It's not affecting me, Just a weird display thing.
>

Will be fixed in next release[1].

[1] https://gerrit.asterisk.org/c/asterisk/+/14924

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-dev] Asterisk 18.0.0 Now Available

[John Kiniston]

Is anyone else seeing menuconfig give the wrong description app_audiosocket
and chan_audiosocket selections with this release?

I've tried on two systems and I'm seeing the same thing, If I highlight
app_audioosocket I get a description of  AST_MODULE_INFO(
and chan_audiosocket has a description of AST_MODULE_INFO(ASTERISK_GPL_KEY,
AST_MODFLAG_LOAD_ORDER,



It's not affecting me, Just a weird display thing.

On Tue, Oct 20, 2020 at 5:02 AM Asterisk Development Team <
asteriskt...@digium.com> wrote:

> The Asterisk Development Team would like to announce the release of
> Asterisk 18.0.0.
> This release is available for immediate download at
> https://downloads.asterisk.org/pub/telephony/asterisk
>
> The release of Asterisk 18.0.0 resolves several issues reported by the
> community and would have not been possible without your participation.
>
> *Thank you!*
>
> The following issues are resolved in this release:
>
> *Security bugs fixed in this release:*
> ---
>
>- [ASTERISK-28589
>] -
>
> chan_sip: Depending on configuration an INVITE can alter Addr of a peer
> (Reported by Andrey V. T.)
>
>- [ASTERISK-28580
>] -
>
> Bypass SYSTEM write permission in manager action allows system commands
> execution
> (Reported by Eliel Sardañons)
>
>- [ASTERISK-28495
>] -
>
> res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash
> (Reported by Alexei Gradinari)
>
> *New Features made in this release:*
> ---
>
>- [ASTERISK-6863
>] -
>
> [patch] allow Asterisk to set high ToS bits as non-root on Linux
> (Reported by Matt Addison)
>
>- [ASTERISK-17491
>] -
>
> CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do
> anything
> (Reported by candrews)
>
>- [ASTERISK-28639
>] -
>
> res_pjsip_endpoint_identifier_ip: Add ability to match on source port
> (Reported by Sean Bright)
>
>- [ASTERISK-28614
>] -
>
> app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only
> "sending"
> (Reported by lvl)
>
>- [ASTERISK-28613
>] -
>
> func_curl: CURLOPT cannot set Content-Type header
> (Reported by Martin Tomec)
>
>- [ASTERISK-28533
>] -
>
> func_jitterbuffer: Add support for video synchronization
> (Reported by Joshua C. Colp)
>
>- [ASTERISK-17808
>] -
>
> [patch] Unregister a realtime moh class
> (Reported by Byron Clark)
>
>- [ASTERISK-28489
>] -
>
> Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI
> domain
> (Reported by Stas Kobzar)
>
> *Bugs fixed in this release:*
> ---
>
>- [ASTERISK-29109
>] -
>
> res_pjsip_session: Asterisk 18 does not progress calls due to codec
> negotiation after upgrading from Asterisk 16
> (Reported by Ross Beer)
>
>- [ASTERISK-25665
>] -
>
> Duplicate logging in queue log for EXITEMPTY events
> (Reported by Ove Aursand)
>
>- [ASTERISK-29043
>] -
>
> app_queue: Leave empty sometimes not recorded as abandoned
> (Reported by Kfir Itzhak)
>
>- [ASTERISK-29042
>] -
>
> res_parking: Parker UUID is no longer copied
> (Reported by Misha Vodsedalek)
>
>- [ASTERISK-28878
>] -
>
> chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
> (Reported by Joseph Ades)
>
>- [ASTERISK-29046
>] -
>
> pbx: Deadlock when doing a reload, while simultaneously doing an
> ExtensionState on a pattern match hint that ends up adding an extension
> (Reported by Ramarajan)
>
>- [ASTERISK-29040
>] -
>
> res_speech: Assertion on format
> (Reported by Nickolay V. Shmyrev)
>
>- [ASTERISK-29001
>] -
>
> chan_pjsip does not process or forward 181 responses
> (Reported by Torrey Searle)
>
>- [ASTERISK-29034
>] -
>
> Lastpause of realtime members is reseting
> (Reported by Evandro César Arruda)
>
>- [ASTERISK-27273
>

Re: [asterisk-dev] Problem with SDP session id in 200 OK during ReInvite

[Michael Maier]

Hello Joshua,


On 27.10.20 at 10:07 Joshua C. Colp wrote:
> On Mon, Oct 26, 2020 at 2:02 PM Michael Maier  wrote:
> 
>> Hello!
>>
>> I'm facing the problem, that *sometimes* the SDP session ID isn't
>> incremented in the 200 OK, which asterisk sends as answer to a ReInvite it
>> got from the peer (use case: session
>> timer handling). This leads to broken calls, because the SDP session ID
>> must be incremented if the session description has changed (the session
>> description has changed).
>>
>> Modifying the SDP session ID is possible in
>> res/res_pjsip/pjsip_message_filter.c / filter_on_tx_message() for SDPs
>> contained in an Invite, which is created and sent by asterisk.
>> At the moment, I'm already modifying the SDP session ID at this place,
>> because of another problem:
>>
> 
> 
> 
> 
>> Maybe it's too late and the processing of the 200 OK doesn't hit this part
>> at all?
>>
>> Or is there any other possibility to modify the SDP session ID contained
>> in the 200 OK, that is sent by asterisk as an answer to a ReInvite?
>>
> 
> It should be modifiable there, it injects itself at the transaction layer
> to be called for both requests and responses. I'm not sure under what
> scenarios (if any) it would not be called.

thanks for your estimation! Meanwhile, I hope I have found the problem now. I 
changed my workaround to always statically modify the SDP session id (the 
condition seemed to be the
problem). But tests are going on.


Thanks
Michael

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[asterisk-dev] Asterisk 13 and Asterisk 17 Security Fix Only

[Joshua C. Colp]

Greetings all,

This is a reminder that as of October 24th Asterisk 13 has gone security
fix only. This means that it will receive no new bug fixes, and any
outstanding bug fixes which have been merged (or are up for review) will be
released in an upcoming final bug fix release. Asterisk 13 has been with us
a long time - since October 24th of 2014, but its time is now over. If you
have not yet made plans to upgrade to either Asterisk 16 or Asterisk 18
then now is the time to start.

As well if you are an Asterisk 17 user then as of tomorrow (October 28th)
it too will enter security fix only status.

A great resource for being aware of this information and planning is the
Asterisk Versions[1] wiki page. Timelines are provided on each Asterisk
version for when it transitions to security fix only and ultimately end of
line.

>From a developer perspective this means we no longer need to cherry pick
changes to Asterisk 13 and as of tomorrow Asterisk 17. We'll work through
any outstanding reviews which are active and then clear out the remaining.

Cheers,

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-dev] Problem with SDP session id in 200 OK during ReInvite

[Joshua C. Colp]

On Mon, Oct 26, 2020 at 2:02 PM Michael Maier  wrote:

> Hello!
>
> I'm facing the problem, that *sometimes* the SDP session ID isn't
> incremented in the 200 OK, which asterisk sends as answer to a ReInvite it
> got from the peer (use case: session
> timer handling). This leads to broken calls, because the SDP session ID
> must be incremented if the session description has changed (the session
> description has changed).
>
> Modifying the SDP session ID is possible in
> res/res_pjsip/pjsip_message_filter.c / filter_on_tx_message() for SDPs
> contained in an Invite, which is created and sent by asterisk.
> At the moment, I'm already modifying the SDP session ID at this place,
> because of another problem:
>




> Maybe it's too late and the processing of the 200 OK doesn't hit this part
> at all?
>
> Or is there any other possibility to modify the SDP session ID contained
> in the 200 OK, that is sent by asterisk as an answer to a ReInvite?
>

It should be modifiable there, it injects itself at the transaction layer
to be called for both requests and responses. I'm not sure under what
scenarios (if any) it would not be called.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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