similar to: DPMA 3.5.5 Released

El 06/09/13 14:44, Malcolm Davenport escribi?: > Howdy, > > Please forgive the off-list e-mail. I'm not subscribed to the list, I only peruse the archives. > > The follow up from George is correct. For phones that have already been attached to DPMA, DPMA disables the enable_check_sync phone setting. > > For phones that have not yet been attached to DPMA, that's not

Hi, I'm just trying to install the DPMA on my Asterisk. I already made the upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: *mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 * *compiling Asterisk-Cert2 1.8.11* *./configure make make install make config * Afther that i register the DPMA license, and finally copied the * res_digium_phone.so* to

Looks like res_digium_phone will need some work for Asterisk 12... WARNING[9372]: loader.c:561 load_dynamic_module: Error loading module 'res_digium_phone.so': /usr/lib64/asterisk/modules/res_digium_phone.so: undefined symbol: __ao2_container_alloc

Greetings all, After a long day of fighting with GTalk and having it finally working, I wanted to setup DPMA on my Digium phone. So first of all, I had to reinstall it all and reconfigure it all, since it works only on certified versions, and my installation was not from the certified branch. It took a long time of recompiling, testing, adding missing stuff, but I got it straight. Now, I

During testing with the Digium phones I have run into a problem where I try to make a change to the sip device name. I make the device name change in sip.conf then make the matching change to the lines in res_digium_phone.conf. I then do 'sip reload' and 'module reload res_digium_phone.so'. I then end up with phones that I cannot bring into service no matter what I have tried. They

Consider the following scenario: 1) One or more Digium DPMA phones are plugged into the network. I know their IP addresses and MACs. 2) The Asterisk I want to use as the telephony server starts without the DPMA module. Therefore there are no DPMA sessions between the phones and the server. 3) I install DPMA on the server, and write its configuration file for the phones. I will tie each phone to

Is there a way to use DPMA to enumerate the Digium phones that are plugged in and visible in the local network, but not (yet) configured through the DPMA configuration files in Asterisk. I would like to write a frontend that lists the DPMA capable phones, presents a GUI to specify the various options, then write the configuration files as required and make the phones read these settings. Ideally

Asterisk 12.5.0 DPMA 12.0_2.0.0 Ubuntu 12.04 64 bit [2014-08-21 16:37:49] WARNING[5797]: phone_users.c:5236 set_and_process: User SIP settings missing or invalid I'm getting the error message above when DPMA is enabled, using a fresh build but with my config files from Asterisk 11. Any idea what it means? I can't find the "phone_users.c" file to examine the source

Asterisk 12.5.0 DPMA 12.0_2.0.0 Ubuntu 12.04 64 bit WARNING[5797]: presencestate.c:147 ast_presence_state_helper: No provider found for label CustomPresence ERROR[5797]: pbx.c:4375 ast_func_write: Function PRESENCE_STATE not registered I only see these when DPMA is enabled. Any ideas what causes this or how to correct it? -- Mitch

I have the following scenario. A small network has DHCP but does not publish option 66. An Asterisk server is on the network, but the Asterisk version does not support DPMA and it is hard to switch the version. However, there is a possibility to have a web server and an mDNS (Avahi) server. I have been reading about provisioning Digium phones without DPMA, and it mentions that option 66 can

On Fri, May 1, 2015 at 10:43 AM, Robert Broyles <robert at webservicesaz.com> wrote: > We love our Digium phones and DPMA - but we really need it to work on our > Realtime Platform. Otherwise we lose all the cool features and they are > just standard SIP phones. > > Anyone working on a solution for this? Or anyone from Digium see this on > the roadmap? > Hey Robert -

I read somewhere that DPMA is not supported for Asterisk 12. Can anyone confirm or deny that? If not supported yet, will it be? If so, when? -- Mitch

We love our Digium phones and DPMA - but we really need it to work on our Realtime Platform. Otherwise we lose all the cool features and they are just standard SIP phones. Anyone working on a solution for this? Or anyone from Digium see this on the roadmap?

> > Apparently notify.check-sync does work but only if you're NOT using > the DPMA. I just tried it and the phone just responds with a 200/OK > and does nothing. Did you disable enable_check_sync in the xml config? By default this option is enabled and phones should restart with check-sync SIP NOTIFY

The Asterisk Development Team would like to announce the release of Asterisk 16.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.1.0 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:

Ok , digging more into this i could see that (timers=no) and (timers=forced) not work asterisk not pay attention to this options when is reloaded cli not say anything and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint. I wonder to force asterisk to refresh the session in some cases when is needed . pjsip is able to

The Asterisk Development Team would like to announce the release of Asterisk 13.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.23.0 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 15.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.6.0 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:

On Tue, Nov 29, 2016, at 07:15 AM, Barry Flanagan wrote: > On 29 November 2016 at 10:56, Jonathan H <lardconcepts at gmail.com> wrote: > > > Any ideas why a VPS, cloned from another instance (DigitalOcean > > "droplets" if it matters), won't run on the new instance? > > > > Everything else is the same; region, memory, disk, hypervisor version etc.

Joshua Colp wrote: > Michael Str?der wrote: >> Joshua Colp wrote: >>> Michael Str?der wrote: >>>> 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