similar to: DPMA - Asterisk Realtime

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 -

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

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

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

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

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

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

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

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

Greetings, We don’t normally announce DPMA releases on here but 3.5.5 was just released which resolves a compatibility issue between the latest versions of Asterisk (using PJSIP 2.10) and DPMA. For TCP or TLS traffic it was possible for a crash to occur. It’s recommended to update to 3.5.5. Installs which select DPMA in “make menuselect” will automatically get the latest version or the tarballs

> > 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

Hi, So I'm using Asterisk Realtime Queues and Queue members on 1.4.28. I've noticed if there are no people in the queue when a call enters, even after a queue member enters, the call is never rang to him. From the debug, it seems that Asterisk is only grabbing the queue member list upon entering the queue. And not again until another call enters the queue. As a result, a caller will

I have been looking for the specs (format, bit rate, ect) on custom ringtones for digium phones. Using the DPMA how would I deliver the ringtone to a digium phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120820/cb4927d0/attachment.htm>

Hello, I've recently given a try to a Digium D70 phone. At the moment, I'm configuring them though config files with a DHCP server and not using DPMA. Of course, I'm connecting them to Asteris (PJSIP stack on 13.7.0). Which is the best place to: - read about past issues - open new tickets for remaining issues. Best regards -------------- next part -------------- An HTML attachment

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

I'm wondering if there's a way to set which periodic-announce file is played from my dialplan, much like setting the monitor-filename. Something like this: exten => s,n, Set(PERIODIC_ANNOUNCE=foo) This would be a great feature if it doesn't already exist. Or perhaps there's a better way to do this. Thanks for your time. -- Regards, Robert Broyles

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

It's my understanding that the backport is available now in 1.4. However, seem to be having some issues with it. Just wondering if I have everything setup right. I'm running 1.4.26.2 realtime. queue_members: `uniqueid` int(10) unsigned NOT NULL auto_increment, `membername` varchar(40) default NULL, `queue_name` varchar(128) default NULL, `interface` varchar(128) default NULL,

Is there a way to set "na.rm=TRUE" as a default, so that this does not have to be re-specified for all of the functions requiring this option? Thanks in advance, Stephanie Broyles sbroyl@lsuhsc.edu [[alternative HTML version deleted]]