Q: If I have a suggestion for a new feature for Digium phones, whom should I contact?
A: Send an e-mail to firstname.lastname@example.org. E-mails received there are read and reviewed by members of Digium's product management team. We may not be able to respond to every e-mail, but all are read.
Q: Why do I need to run a new / different version of Asterisk to use Digium phones?
A: The phones should operate normally with any modern version of Asterisk from 11 forward. 1.8 and 10 releases required the special "-digiumphones" branches in order to get User Presence, Visual Voicemail, Call Parking, Call Queues and enhanced provisioning, but could function as standard SIP phones otherwise.
Q: Why is DPMA not an open source module?
A: Keeping the protocol and implementation proprietary prevents counterfeit or "clone" phones. Digium has put a great deal of time, effort and money into developing both Asterisk and our phones. We freely give away our efforts on Asterisk. We use revenue from the phones to help fund the ongoing development of Asterisk. Thus we need to make sure that the phones remain a viable business.
Q: What kind of support is available from Digium for phones used with Asterisk? Do I need paid Asterisk Support from Digium?
A: Digium supports the use of Digium phones with Asterisk. You don't have to have paid Asterisk Support in order to get help with the phone. The included support doesn't mean Digium will help you upgrade Asterisk, or answer questions about why calls to your service provider don't work, etc.
Q: Does Digium have a demo unit program? I would like to test out a phone.
A: Demo pricing is available to Registered and Select level Digium reseller partners. If you're an integrator, reseller, solution provider or OEM who is not currently part of our channel program, please contact our Asterisk Integrator recruiter who can help you sign up as a reseller.
Q: When the phones register with Asterisk, is the registration information encrypted?
A: Information related to the provisioning of the phone as well as information specific to applications running on the phone is encrypted. Standard SIP signaling, e.g. call setup, dialogs, etc. is encrypted for D6x model phones beginning with phone firmware 184.108.40.206.
Q: Do the phones support IAX2?
A: No. Digium's phones are SIP phones. IAX2 was built as a trunk-side protocol (thus the "Inter Asterisk eXchange" name) and lacks many of the features / capabilities that are required for a desktop phone.
Q: Is there Bluetooth headset support?
A: Digium's D40, D50 and D70 do not directly support Bluetooth connectivity. You can, however, use an external Bluetooth headset connected to the RJ-9 headset jack on the phones. The D65 model phone does support the use of such headsets.
Q: Do the phones display the status of other users on the speed dial buttons?
A: Yes. If you make use of Switchvox or DPMA, then user presence can be seen as a series of presence icons in the phone's Contacts application, and, for the D70, on its secondary LCD. Without Switchvox or DPMA, the BLFs status is limited to generic device status - ringing, on call, on hold - as with any standard SIP phone.
Q: Will the phones support CDP to get voice vlan tags from a Cisco switch?
A: No. Rather than supporting CDP, we support LLDP for switch-controlled vlan assignment.
Q: Do they support AAC-LC?
A: No. The supported codecs for voice calling are listed on the Digium Phones Codecs page. Ringtones are provided in 16kHz raw signed linear format.
Q: Do the phone handsets comply with the HD voice standards (physically)? There are certain parameters that were set up to ensure that the speaker and mic and cavity in a handset would be able to provide good enough quality to utilize the G.722 codec properly.
A: Yes. The speakers (handset and phone), microphones (handset and phone) and the phone itself are designed to support and were tested against audio frequencies sampled by G.722 as well as 16kHz signed linear.
Q: Is it possible to add contacts from the phone and have them stored on Asterisk?
A: We recommend that administrators wanting their users to have control over contacts and/or Rapid Dial keys should provide those users with some tool (that the admin builds, with proper permissions) for managing those contacts.
Q: Do the standard phone features work with the current mainline releases of Asterisk?
A: Yes. Without Switchvox or DPMA, the phones operate like other SIP phones. With Switchvox or DPMA, the phones gain additional capabilities.
Q: Does the voicemail application work out of the box on Asterisk?
A: Yes. If DPMA is used, meaning that the phones are provisioned by it, and the phones are setup on a SIP peer that has voicemail, then the voicemail application works "out of the box."
Q: Can the contact directory be integrated with Exchange global address book?
A: The phone retrieves contacts via XML-formatted files. If one can extract an Exchange address book into properly formatted XML files, then the phone's contacts could match. There's also a user-built (by Tom De Moor) application for LDAP contacts that is available here.
Q: How do you change what soft buttons are enabled or not? Not seeing it in the web UI or in AsteriskNOW. (I.e. how do you control what soft keys appear in various states.)
A: Key mappings are documented, beginning with the XML Configuration 220.127.116.11+ page.
Q: Is there an app development kit?
A: Documentation is available at Phone API Documentation. You'll need firmware 1.3 or greater on your phone in order to take advantage of it.
Q: Do the integrated features only work with Switchvox?
A: Currently phones using DPMA have nearly the same functionality as phones when used with Switchvox - the difference currently (as of DPMA 1.3) being that the Queues application in Asterisk doesn't have the same statistical information as the Queues application in Switchvox, so some fields in the Queues app on the phone will always be blank.
Q: What kind of stock ring-tones and distinctive ringing do the phones support?
A: The phones provide a range of built-in ring-tones. Ringtone selection is controlled via Alert-Info SIP headers as noted in the DPMA Users Guide.
Q: Can you use custom .wav file ring tones?
A: You can use custom ring tones, but you'll need to convert them to raw signed linear files first. For tips on that see Digium Phones Ringtones
Q: Will the phones only work with Asterisk, or will they work with any phone system that supports the SIP protocol?
A: They phones are SIP phones, so they should work with anything that supports SIP. Digium does not do any testing efforts of Digium phones with SIP proxies or user agents that are not Asterisk.
Q: Can we use device status from Lync 2010?
A: The best answer for this question is a question itself: can you get device status from Lync 2010 into Asterisk's dialplan? If so, then the answer is yes. If not, then the answer is no.
Q: Are there any reference configuration files for res_digium_phone.conf available without doing an AsteriskNOW install?
A: Yes. Please see other sections of this Wiki
Q: How can you reload res_digium_phones.conf without doing a restart?
A: From the Asterisk CLI do: "module reload res_digium_phone.so"
Q: Is there a digital "lifter" (a.k.a. electronic hook switch) for wireless headsets?
A: EHS support for Plantronics and Jabra IQ devices is available, beginning in phone firmware 1.1.
Q: Do the phones have a VPN client built in?
A: Yes. Digium's D6x series of phones, beginning with firmware 18.104.22.168, provide support for OpenVPN configuration. See - Digium Phones and OpenVPN for old firmware
Q: When will a version with support for add-on sidecars come out?
A: The D70 and D65, with their capability for pages of contacts, preclude the need for an add-on sidecar.
Q: Is there a way to disable the SIP transfer method and force the phones to use Asterisk's "feature code" transfer method (presumably to force the call to go through the dial plan and trigger additional events or processing?
A: Limit the number of calls to the device to 1 using Asterisk's dialplan; hangup any additional call attempts.
Q: Is HDVoice a paid codec?
A: No, G.722 is not a royalty bearing codec and does not require any additional license fees by phone users.
Q: If I reboot the phone, will the switch reset?
A: Yes. If the phone is made to reboot, for example, a firmware upgrade, the switch will reset and the chained device will lose its connectivity until the reboot cycle is nearly complete. Instead, if the phone is just reconfiguring, to update contacts or lines for example, the switch remains active.
Q: In what cases does the phone need to reboot?
A: The phone needs to reboot to change network configurations, update firmware, change log levels, and for certain codec configuration changes.
Q: Is Digium planning to open source the firmware for the phones?
Q: What protocols are supported for non-DPMA provisioning of a Digium phone?
A: As of the 1.0.x firmware releases, HTTP and HTTPs are supported. The 1.1 firmware provides support for FTP and FTPs. TFTP is not supported.
Q: Do Digium phones support multiple call appearances?
A: Yes. Digium phones support multiple call appearances. All call appearances associated with a particular SIP account are tied to that line. Each call appearance will show up in the call field and can be navigated using the arrow keys.
Q: Can Digium D40, D50 and D70 model phones display Kanji characters?
A: No. Latin characters within the UTF-8 space can be displayed, but Kanji symbols / characters cannot.
Q: What are the MTBFs for the various phone models?
A: D40 - 98.4 years, D45 - 110 years, D50 - 92.4 years, D60 - 111.6 years, D62 - 109.6 years, D70 - 79.5 years, D80 - 72.9 years.
Q: When upgrading phone firmware, my phone says something about a failure to write to disk. What's going on?
A: Your firmware download may have been incomplete and you may be attempting to apply corrupted firmware to your phone. Please see the MD5 Sums on the Firmware page and compare those to your downloaded files.