[asterisk-dev] Asterisk 13.15.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of
Asterisk 13.15.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.15.0 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:

*New Features made in this release:*
---
- [ASTERISK-26878 ]
- func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)
- [ASTERISK-26863 ]
- res_pjsip: Add endpoint identification scheme based on a configured SIP
header/value
(Reported by Matt Jordan)
- [ASTERISK-17428 ]
- [patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)

*Bugs fixed in this release:*
---
- [ASTERISK-26851 ]
- res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26897 ]
- chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
- [ASTERISK-26916 ]
- res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-26705 ]
- libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-26850 ]
- res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)
- [ASTERISK-26484 ]
- res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
argument.
(Reported by Vinod Dharashive)
- [ASTERISK-26776 ]
- res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)
- [ASTERISK-26880 ]
- Asterisk crashes when multiple speex users join confbridge with pp_vad
and dtx enabled
(Reported by Kirsty Tyerman)
- [ASTERISK-26862 ]
- app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)
- [ASTERISK-26732 ]
- res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)
- [ASTERISK-26879 ]
- PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
- [ASTERISK-26867 ]
- autochan: Locking in a function ast_autochan_destroy() on destroyed
channel (after masquerade).
(Reported by Krzysztof Trempala)
- [ASTERISK-26869 ]
- res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s
extension
(Reported by Torrey Searle)
- [ASTERISK-26668 ]
- core: Malformed pattern matching extension (various factors) results in
crash
(Reported by xrobau)
- [ASTERISK-26865 ]
- chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)
- [ASTERISK-26872 ]
- Bundled pjproject fails to build when tarball downloaded with curl due to
md5 verification failure in Docker containers (or when there is no terminal)
(Reported by Matt Jordan)
- [ASTERISK-26717 ]
- Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)
- [ASTERISK-26643 ]
- Extra new line in Device field of DeviceStateChange AMI Event after
restart of Asterisk
(Reported by Roman Bedros)
- [ASTERISK-25237 ]
- stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)
- [ASTERISK-26857 ]
- chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
- [ASTERISK-26841 ]
- chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26822 

[asterisk-dev] Fwd: Asterisk 14.4.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of
Asterisk 14.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.4.0 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:

*New Features made in this release:*
---
- [ASTERISK-26878 ]
- func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)
- [ASTERISK-26863 ]
- res_pjsip: Add endpoint identification scheme based on a configured SIP
header/value
(Reported by Matt Jordan)
- [ASTERISK-17428 ]
- [patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)

*Bugs fixed in this release:*
---
- [ASTERISK-26851 ]
- res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26897 ]
- chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
- [ASTERISK-26916 ]
- res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-26705 ]
- libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-26850 ]
- res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)
- [ASTERISK-26484 ]
- res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
argument.
(Reported by Vinod Dharashive)
- [ASTERISK-26776 ]
- res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)
- [ASTERISK-26880 ]
- Asterisk crashes when multiple speex users join confbridge with pp_vad
and dtx enabled
(Reported by Kirsty Tyerman)
- [ASTERISK-26862 ]
- app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)
- [ASTERISK-26732 ]
- res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)
- [ASTERISK-26879 ]
- PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
- [ASTERISK-26867 ]
- autochan: Locking in a function ast_autochan_destroy() on destroyed
channel (after masquerade).
(Reported by Krzysztof Trempala)
- [ASTERISK-26869 ]
- res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s
extension
(Reported by Torrey Searle)
- [ASTERISK-26668 ]
- core: Malformed pattern matching extension (various factors) results in
crash
(Reported by xrobau)
- [ASTERISK-26865 ]
- chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)
- [ASTERISK-26872 ]
- Bundled pjproject fails to build when tarball downloaded with curl due to
md5 verification failure in Docker containers (or when there is no terminal)
(Reported by Matt Jordan)
- [ASTERISK-26717 ]
- Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)
- [ASTERISK-26643 ]
- Extra new line in Device field of DeviceStateChange AMI Event after
restart of Asterisk
(Reported by Roman Bedros)
- [ASTERISK-25237 ]
- stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)
- [ASTERISK-26857 ]
- chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
- [ASTERISK-26841 ]
- chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26822 

Re: [asterisk-dev] Asterisk-11 DTMF bug ?

[Mark Murawski]

On 4/6/17 4:23 PM, Matt Fredrickson wrote:


Hey Mark,

First off, thanks for reaching out to the Asterisk community to talk
about the trouble you're having :-)

This actually is a development related mailing list - so primarily for
development of Asterisk C source code level discussions, project
policy related discussions, and sometimes protocol development related
discussions.

Your question appears to be configuration or at the very least debug
related, rather than development related.  I would try referring it to
the asterisk-users mailing list or the community.asterisk.org forums
instead.

I think you probably should add more data about your Asterisk
configuration in this case, as well as your expectations as to what
should be working and what is not working, as that is not clear to me
from your description.  Are you not seeing DTMF bridged between the
two parties in the call, or is some other DTMF triggered behavior you
expect Asterisk to be demonstrating not occurring?  These are
questions I would hope to be answered when posting a follow up on one
of the other lists.

Hope that helps, and best wishes to you in resolving your problem! :-)



Hi Matt,

As a long time community member, I do understand the split of -users and 
-dev.  This is an internal implementation question/discussion and I 
believe this is more suited to the -dev list.


I'm in the process of collecting more test cases, so far I've also 
reproduced the issue with a soft phone.  So it does appear that it's not 
dependent on the endpoint user agent to exhibit this problem.


The purpose of this post was to put out feelers to see if this was 
possibly a known issue with handling DTMF with a B2BUA scenario. While 
it may wind up being a configuration issue to fix this, the expectation 
is that with the current settings, DTMF should 'just work'.  The 
endpoint is clearly sending RFC2833, and asterisk is not interpreting it 
in the B2BUA flow.


New Details: If I switch the endpoint to SIP INFO, then asterisk does 
interpret the DTMF.  To me, this signals that this is probably a bug.  
Given all else being equal that the asterisk config has not changed, and 
switching from 2833 to info, fixes the issue.


I'll be drawing out some diagrams and describing some test cases within 
a day or two.



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Re: [asterisk-dev] asterisk-doc and asterisk-gui mailing list

[Matt Fredrickson]

On Fri, Apr 7, 2017 at 7:56 AM, Andrew Latham  wrote:
> seeing spam on these lists: asterisk-doc and asterisk-gui
>
> Can some one have a look?

We'll check into it and see what's going on.  Thanks for catching this.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-dev] asterisk-doc and asterisk-gui mailing list

[Andrew Latham]

seeing spam on these lists: asterisk-doc and asterisk-gui

Can some one have a look?

-- 
- Andrew "lathama" Latham  -
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