similar to: Asterisk 16.13.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 16.13.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.13.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 13.36.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.36.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 13.36.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.36.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 17.7.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.7.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 17.7.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.7.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 13.37.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.37.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 16.14.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.14.0 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:

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:

The Asterisk Development Team would like to announce the release of Asterisk 17.8.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.8.0 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:

Hello, I found a problem with latest Asterisk 16.13.0 and SNOM3xx phones (EOL) running the above FW. Incoming calls are no more working, we get error 404 despite the fact that broken registrar is on for the account. Previous FW for this phones don't have the problem. At this time I open a ticket at SNOM support, don't know if I should also open an issue at jira. Regards -- Daniel

HI! I'm trying to compile asterisk 13.8.2+ on openSUSE Linux but it fails. It seems file ./main/libasteriskssl.so.1 is present when it fails. Building 13.7.2 works without any problem. It fails since 13.8.0. $ ./bootstrap.sh $ ./configure $ make menuselect.makeopts;menuselect/menuselect --enable chan_ooh323 $ make .. failure (see message below) Any hint is appreciated. Thanks in advance.

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

asterisk-16.13.0-rc2. Fedora 32 pjsip won't load because of undefined symbols: [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_aor.so': /usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module

With the current CVS-HEAD line 88 of app_rxfax.c causes an error. #if (ASTERISK_VERSION_NUM <= 010300) chan->callerid, app_rxfax.c:88: error: 'struct ast_channel' has no member named 'callerid' Commenting out the if else combination of course gives a clean compile. -- Dave Cotton <dcotton@linuxautrement.com>

On 05.09.20 at 15:22 sean darcy wrote: > asterisk-16.13.0-rc2. Fedora 32 > > pjsip won't load because of undefined symbols: This means, that your pjsip library doesn't match the asterisk binary. It's best to remove the independent pjsip library and compile asterisk[1] with the bundled pjsip library. Doing it this way ensures that pjsip and asterisk match for sure (and some

16.13.0, pjsip I'd like to get an alert if a call fails to authenticate: if "Failed to authenticate" then mail someone the source ip endif As I look at ami or ari, they deal with calls in channels. Is there a way to get failed invites or registers ? sean

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

On Wed, Sep 30, 2020 at 9:06 AM sergio <sergio at outerface.net> wrote: > On 30/09/2020 14:59, Joshua C. Colp wrote: > > > latest version of 16 on Ubuntu > > 16.12.0~dfsg-1 ? > I don't use packages. I built a fresh copy of 16.13.0 as it is latest and used that. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and