similar to: Asterisk 12 and DPMA

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

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

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

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?

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

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

I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION ISOLATION LEVEL READ COMMITTED 2018-12-04T16:29:27.254384Z 229 Prepare SELECT * FROM queues WHERE name = ? 2018-12-04T16:29:27.254902Z 229

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

Asterisk 12.5 I have a reproducible segfault using the MeetMe application. How do I gather the necessary information (backtrace, core dump...) to submit a bug report? -- Mitch

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 -

Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the

> > 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

Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.

Asterisk 12.4 I am seeing message "Error opening file for reading: Permission denied" several times during the asterisk startup (asterisk -cvvvvv) but it doesn't say which file. Is there a way to find out which file is having trouble? -- Mitch

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>

Converting this customer from a MiTel system to asterisk. Discovered that the inbound calls from the T1 are going to extension 366. (This was mapped in the MiTel for some arcane purpose.) The dial plan I am currently using is shown below. When loading the dial plan, I get this warning: WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for an extension is strongly

The "core show channels verbose" command shows a "calls processed" value. Mine is currently 1928273. Exactly what does this figure represent? How is a "call" defined in this context? -- Mitch