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  • Please make sure you read ALL of the information below before installing the software on your device.
  • Please install and test the software in your environment before mass deployment
  • We encourage you to read and follow our security advisories.
  • This release does NOT directly include the OpenVPN feature in firmware file anymore. If you wish to use this feature please make sure you understand all security implications and follow this link after installing the software.

Release notes

New Features

  • SAP-271 ADD: On Snom 710 / D710, 715 / D715, 720, D725 - Messages (IM) are also displayed & updated in Connected State. If messages are received, they now take precedence in idle & connected, plus support for auto-scrolling of long messages.
  • SCPP-6799 ADD: TR-069 Base64 encoding of :X_000413_GetWebContent data in response. Implementation of a general base64 encoding for all X_000413_GetWebContent requests, - not only for image or binary data but for all data returned.
  • SAP-65 Epic: Built-in Bluetooth support in Snom D765, see: supported / approved BT headset list
  • NONE ADD: RTCP-XR voice quality collector can be configured per identity, plus it can be decided if the corresponding outbound proxy should be used for the collector or not
  • SCPP-5406 ADD: Circular diagnostic logging on USB stick (SD card in MP). Allows continuous recording of PCAP, SIP-Trace and SYSLOG (free-selectable) to USB, plus option for a pass phrase to encrypt the files using an AES 256-bit CBC cipher, see Category: USB
  • NONE ADD: If the syslog on USB stick (SD card in MP) feature is enabled, the log level can be set via a file named syslog.lvl on the USB stick. The phone reads the first byte from the file and uses the values from '0' to '9' for the loglevel, other values are ignored. See USB-SYSLOG
  • NONE ADD: Firmware update possible via USB stick (SD card in MP). See USB-FW-Update
  • SCPP-4507 ADD: New timezones for Colombia and Ecuador added.
  • SCPP-6582 ADD: Support for Greek dial tone scheme


  • SAP-222 UPD: Support for Public CAs: Iden Trust (issuer of: Let's Encrypt), Baltimore Cyber Security (issuer of: Verizon Public SureServer), GoDaddy. (few CA's expiring 2015 are removed)
  • SCPP-6472 UPD: Support only modern Cipher Suites in Server Side TLS. TLS 1.2 ciphersuites reduced from 90 to 42 oriented towards Mozilla Intermediate (Note: more details planned for documentation)
  • SCPP-6400 UPD: Remove RC4 ciphers from OpenSSL stack (see RFC 7465, note: NULL ciphers are already disabled by default on OpenSSL stack)
  • SAP-112 FIX: Extended security check on http_proxy and vpn_netcatserver setting. (Not mandatory, if already taken into account: How_do_I_secure_my_phone? )
  • SAP-158 FIX: Security.htm (http://phone-IP-address/security.htm) web user interface page did not show up after settings reset
  • SCPP-6353 FIX: Unable to logout from the WUI (web user interface) in specific case if 'admin_mode_upon_http_login' is enabled
  • SCPP-6300 FIX: LLDP: phones did not defend / detect IP in its own VLAN. In networks featuring Proxy ARP, ARP packets transmitted over the whole network, the LLDP VLAN managed Snom IP conflict detection did not work.
  • SAP-129 FIX: Align DHCP IPv6 (DHCPv6) handling with default of dhcp_v6 - off --> IPv6 switched off completely
  • NONE: REV: Re-enable ethernet link detection for snom MP by default
  • SAP-104 UPD: Default value of ip_frag_enabled should be on --> OOB support for IP fragmentation. Recommended read: IP frag
  • SCPP-6878 FIX: Default value of ip_frag_enabled (on , see above SAP-104) was not initially applied. A workaround till the fix was: changing the setting to off then back to on and then rebooting the phone.
  • SCPP-6358 FIX: Snom 710: 802.1x EAP pass-through is not working using VLAN setup via LLDP
  • SCPP-5810 FIX: Snom 710: 802.1x EAP EAPOL Start packet received on the LAN port should be discarded and not forwarded
  • SCPP-6343 FIX: Snom 7x5, D7x5 & D3x5: 802.1x EAP pass-through is not working using VLAN setup
  • NONE FIX: Prevent additional link-down on snom 720 & 760 in advertising gigabit ethernet
  • SCPP-6753 FIX: TR-069: HTTP authentication - must submit lower case according to RFC 2617 for realm in str("REALM=\"")
  • SCPP-6440 Lost support for path in DHCP option 66, due to aligment with RFC 2132 (DHCP option 66 contains the IP address or the hostname). All redirection & provisioning solutions, distributing the path and maybe file-name info via DHCP option 66 (e.g. cannot use option 67), (irrelevant of 43 encapsulation is used or not). Implemented a back- & forward-compatible handling for 66 / 67, covering RFC 2132 and none-RFC 2132 e.g. 66 with path and with/without file name
  • SCPP-6646 Redirection and/or provisioning are broken as phones stop replacing the {mac} system variable in a given path e.g.{mac}/snom.htm since curly braces are percent-encoded. All redirection & provisioning solutions using the {mac} variable in paths e.g. provided by DHCP 66/67 (irrelevant of 43 encapsulation is used or not). Stop percent-encoding for variable / reserved characters, here esp. the curly braces.
  • SAP-90 XML phonebook feature (e.g. via Action URL) is broken by percent-encoding of e.g. Semicolon, Equal sign, or other reserved characters, see RFC 3986. Basically all solutions that rely on Action URL's with reserved characters like: ; or = for example. WebUri path segments will be encoded as specified by RFC 3986.
  • SCPP-6368 FIX: Provisioning of backlight settings (use_backlight, backlight, backlight_idle) are not effective immediately
  • SCPP-5840 FIX: TR-069: Several Parameters are rejected
  • SCPP-6288 FIX: TR-069: On new bootstrap without REBOOT TR111 Gateway info lost
  • SCPP-6662 FIX: Align rescue mode factory reset with factory reset performed by web / phone user interface (ensuring factory-defaults)
  • SCPP-6884 FIX: Busy Lamp Fields stop working / BLF SUBSCRIBE closed after periodic provisioning. The phone sends an un-SUBSCRIBE (Expire: 0) after periodic provisioning / setting refresh. Observed only if BLF value was just ext.- / user- part e.g. 123. Provisioning a full SIP-URI e.g. sip:123@ or sip:123@hostname.local worked as expected. Fixed by the corrections in SCPP-6579 below.
  • SCPP-6806 FIX: In transfers / attended transfers the Caller ID is not updated if the UPDATE info is given in P-Asserted-Identity. The transfers work, but display continues to show initial connection info.
  • SCPP-6263 In case a re-invite doesn't contain the direction attribute, the phone was sending incorrect Connection Information (c): IN IP4 & port: 0 in SDP. E.g. Cisco Call Manager v.10.0. and feature: "Play music during hold:" in "Advanced Settings-Audio" is activated. The CCM responds with an "inactive" to the SDP info above, leading to no MoH on first Hold, but works on second attempt. Implemented a new setting: remote_3264_hold which is on per default to support a remote side supporting RFC 3264 style hold when a=sendrecv is missing in SDP. To keep the Hold signaling as specified in RFC 2543 just turn it off.
  • SCPP-6678 The hash / number sign: # is submitted percent-encoded (escape characters) if configured or provisioned on e.g. BLF Fkey or Ext. Example symptom: BLF pick up and speed dial e.g. can be broken. All setups / solutions, configuring or distributing a value to BLF / Ext. / speed dial in a form that includes a # sign, e.g.: 23#45 . In that cases the phone will send an INVITE to URI sip:23%2345@... While this behavior is aligned with RFC 3261 - it may not be supported or maybe handled as an unexpected exception by the systems. UPD - New setting to implemented a back- & forward-compatible handling that can be controlled per identity via: number sign encoding - Default is: on , to be out-of-box compliant with RFC 3261
  • SCPP-6579 & SCPP-6904 FIX: Dialing & subscribing of alphanumerical numbers (values that begin with a letter) in general was broken (via phone and web user interface), e.g phone subscribed to @p123 (or speed dialed just: INVITE sip:p123) if only p123 was entered or provisioned the phone performed an DNS query on the alphanumeric name. By dialing or provisioning a full SIP-URI with the alphanumerical number e.g. sip:p123@ or sip:p123@hostname.local worked as expected.
  • SCPP-6350 FIX: REGISTER request contains the following invalid header: X-Real-IP: {x-snom-adr} , due to missing address substitution 
  • SAP-139 REGISTER requests may fail as they contain the following invalid header: WWW-Contact: <https://%7Bx-snom-adr%7D> due to curly braces being percent-encoded, like in SCPP-6646 (provisioning). SIP-solutions that expect /require a valid WWW-Contact header. Stop Percent-encoding for this header / reserved characters.
  • SCPP-6478 FIX: DTMF broken by loss of DTMF RFC 2833 negotiation in SDP answer 101 telephone-event, if user_full_sdp_answer is off 
  • SCPP-5989 After holding (or attempt to transfer) a call the phone sends a BYE and shuts down the call, due to SDP answers with amount of m lines not fit the offer (e.g. m=video, or two "m=..." lines in SRTP/SAVP). In case of RTP/SAVP:optional & RTP Encryption:on - the phone immediately closes the call. SIP-solutions aren't full RFC 3264 compliant, e.g. video-enabled IP-PBX or SRTP / SAVP activated IP-PBX, others. UPD - New setting allow_mismatched_sdp_answers allows SDP answers whose m lines are a subset of the offer (off by default, RFC 3264 compliant OOB, out-of-box).
  • SCPP-6337 FIX: Transfer / Attended Transfer may fail, as SIP stack does not correctly distinguish URI parameters from header parameters in specific headers (To/From/Contact)
  • SCPP-6743 FIX: incoming 603 Decline is handled as Busy Here, - no 603 code and / or corresponding message in phone display
  • SCPP-6578 Codec G729 (annexb=no) related calls from e.g. a Cisco IP Phone or other equipment (Voip GW, etc.) are aborted by caller, due to missing fmtp SDP line in response. Adding fmtp line to response even if no rtpmap line is present in received SDP.
  • SCPP-6396 FIX: SIP-Subscriptions provisioned / configured via XML Definition get lost
  • SCPP-6534 Caller ID update (SIP INFO) displays the prefix: sip: E.g. FreeSWITCH (see: mod_sofia.c#1702). Implement a back- & forward-compatible handling for invalid To:/From: header used for caller id indication via SIP INFO. No action required if mod_sofia.c#1702 might be updated in the future.
  • SCPP-6517 FIX: SIP IPv6 grammar must contain the mandated delimiters for IPv6 reference (see RFC 5118 & RFC 3261, the IPv6 reference in the R-URI does not contain the mandated delimiters for an IPv6 reference ("[" and "]")
  • SCPP-6476 FIX: Safe transfer fails in certain SIP environments, due to Microsoft (Lync/SfB) header ms-sensitivity=private-no-diversion in Refer-To
  • SCPP-6674 SIP invites via UDP with a certain size can cause hard to manage / handle fragmentation. Networks / solutions using SIP via UDP and facing challenges with fragmentation. Reducing the SIP invite size by removing unused / less common codecs from the default setting (affects new production or F-reset devices - all codecs supported can be re-added, if needed, of course, see Codec and Codec priority, note: page update pending)
  • SCPP-6377 FIX: If servertype = broadsoft, no headers Required: timer or session refresher are sent, just supported: timers
  • SAP-113 FIX: ecma TR/87 uaCSTA behaviour of optional 'prompt' and 'doNotPrompt' was always treated as 'doNotPrompt' (zero-interaction on-speaker-call attempt)
  • SCPP-6383 Voice quality reports (VQReport) broken, due to wrong format and value of the stop timestamp in the RTCP-XR report, Jitter / Packet loss is always reported as 0. Full refit & maintenance of RTCP, RTCP-XR, RX_Pkts_Lost, Remote_Rx_Pkts_Lost, Timestamps, and BYE values
  • SCPP-6714 FIX: Voice quality report (VQReport) is missing in SIP PUBLISH
  • SAP-244 One way audio after hold / unhold was pressed (only FW September 2015 limited release, please consider this version as deprecated see end of page). SIP-solutions with central MoH in receive-only. In recently introduced substream handling (only public version affected is FW the state of substream was not updated (from hold to unhold, rec-only to sendrec). Solved, substream now updated, in FW and higher.
  • SCPP-6316 FIX: Holding reminder heard by connected party, solved on Snom 715/D715,725/D725,D765,D375, still open for 710/820 see: Known issues - SAP-220
  • SAP-184 FIX: On Snom 870 or 821 receiving an Invite, the user had no audible ringing / ringtone was not played on rare occasion.
  • SCPP-5933 & SCPP-6648 FIX: Choppy audio on synthesizer tones, eg. ringing, dial tone, ring back tone, hold (Snom 715/D715,725/D725,D765,D375)
  • SCPP-6258 FIX: DTMF codes stop working ( no DTMF tone will be heard , nor DTMF codes will be sent in RTP) until a reboot is performed
  • SCPP-6680 Static sound during calls in "heavy" Jitter affected scenarios (Snom 720,760 & MP). Jitter buffer tuning and optimization for Snom 720, 760 & MP.
  • SCPP-6397 FIX: No audio or just noise, as phone changes codec from g729 to g726-32 automatically (no signaling), if g726-32 is set before g729 on codec priority list
  • SCPP-6522 FIX: Snom 7x5 / D7x5 no audio or one-way audio with G.729 codec
  • SCPP-6676 SRTP calls only, no audio in a call after approx. 1hr 30min when hold/unhold was pressed. SRTP enforced / optional SRTP solutions. SRTP roll over counter must not set to 0 after hold/unhold
  • NONE FIX: SRTP calls only - with centralized MoH streaming, Transfers (B xfer's A to C) may result in a one-way audio. A can't hear C.
  • SCPP-6318 FIX: Snom 720, 760: Incomplete audio after 3-way conference resume
  • SCPP-6249 FIX: SRTP calls only, Snom MeetingPoint: Distorted audio in a 3-part conference using G.729. MP no longer allows conference with encrypted G.729.
  • SCPP-6553 FIX: Audio indication for call waiting (CW / CWI) stops frequently (until a reboot is performed)
  • SCPP-6528 FIX: Turn off cw_dialtone had no effect. On/Off always results in stuttered dialtone followed by a normal dialtone.
  • SCPP-6524 FIX: Snom 7x5 / D7x5 Bellcore tones fail to properly playout (missing ring splash / short single ring, visual indication only)
  • SCPP-6674 UPD: remove unused / less common codecs from the default setting (see SCPP-6674 in SIP signaling section, plus Codec and Codec priority, note: page update pending)
  • SCPP-6842 FIX: Call history did not reflect transfered calls. While Caller ID in phone display was updated correctly, the call lists remained to the original / initial Caller-ID. E.g. creates trouble for call-back attempts. Now call list for received calls will contain all remote participants with correct duration / time.
  • SCPP-6870 FIX: If the phone is blind transferred and user decides to hang up (go onhook), it did not fully returned to idle state. Instead it remained in edit / enter mode.
  • SCPP-6835 FIX: Manual IP setting input was broken. Entering of device IP address was limited / (pre-) defined by the previous e.g. DHCP given IP. Other setting input contexts e.g Netmask, Gateway, DNS server had identical or similar "pre-masking" issue.
  • SCPP-6519 FIX: Phones showed an unusable Fkey for BLF List / Monitor Call in Connected State. Now its only displayed, if BLF is configured.
  • SAP-273 DEL: Remove F_REC / F_REC(not:Conference) in all defaults for Call Screen Fkeys on connected variants (note: page update pending). Of course can be provisioned, where it is required.
  • SCPP-6844 FIX: Conference (soft key: F_Conference) is moved to page two when handset mode is activ. Appearing +Spkr key scrolled the Conf.On key to next page. Removing F_REC, see before at SAP-273 alleviated the problem.
  • SCPP-6695 FIX: Auto Redial was offered quite delayed and not in handset mode. CoB option took too long to show up, the user might go onhook already. In addition: if handset was used, phone offered the "Edit_Number" state and showed CoB after going onhook. Other corrections in this context: a held call was preventing the auto-redial prompt from being displayed and retrieving a held call would cancel auto-redial.
  • SCPP-6701 FIX: The Icon / Logo URL Tag in MenuItem needs to be ignored by non-graphical devices (e.g. 300, 720, MP) otherwise the display shows the URL(s). Example: <Icon></Icon> in SnomIPPhoneMenu (note: page update pending)
  • SCPP-6828 UPD: Snom 870 presented the old Snom logo in the analog full screen clock. For consistency the new logo is now also shown on this clock.
  • SCPP-6602 FIX: In directory, edit screens and other contexts, the lower part of letter "g or j" (likely others too) was slightly cut off. In addition the sort category name was not perfectly centered (e.g. Title). Now the presentation of sort categories is redesigned to an more efficient list view. The graphic that overlayed the letters lower part in e.g. edit screens was corrected for a better alignment.
  • SCPP-6401 FIX: Wrong date is displayed in certain time zones (NZ, US Pacific, Australia, others). UTC offset was not taken into account for affected time zones.
  • SCPP-6549 UPD: Never show "Provisioning failed" to user via phone user interface (redirection / prov. permanent or temporally out-of-service, or unreachable, users get confronted with warning which they are not responsible for / related to)
  • SCPP-6171 UPD: Remove Network Wizard (ipv6/vlan/... ) after initial reset of the phone. Besides IPv6 is switched off completely (see SAP-129, Network Protocol), users get confronted with network related settings which they are not responsible for / related to)
  • SCPP-6523 & SCPP-6349 Multiple cancels required to leave an XML menu, XML menu closes automatically (Transfer / Blind Transfer) or expected result of XML minibrowser query are available but don't show up (looks like missing display refresh). On http://phone-ip-address/minibrowser.htm?show=STATE you can see the minibrowser-stack growing. Improve control over Minibrowser-stacking (intro of a new unique identifier "document_id" allowed in all SnomIPPhone...-tags except in SnomIPPhoneBatch, documentation pending)
  • SCPP-6867 FIX: The separator / edit dialog restrictions (digits / alpha & amount limits) and inputtype where kept / copied from one edit-screen to another (e.g. Time to Displayname).
  • SCPP-6412 FIX: If DND is activated the DND overlay (symbol & text) is present in call history menu, making it difficult to read.
  • SCPP-6549 FIX: Disconnected screen is indicated too late, showing wrong disconnected party
  • SCPP-6416 FIX: Snom 370: Dialing of hash (#) key doesn't work. Results in space instead of hash.
  • SCPP-6360 & SCPP-6372 Issue: Snom 870: in some menu dialogs incorrectly labeled virtual keys show up, like F_OK and F_CANCEL, F_ABORT
  • SCPP-6371 FIX: Snom 870: Unexpectedly masked dialed number with an asterisk (*) in call screen
  • SCPP-6406 FIX: Snom 870: Function keys F3 does not work for XML Minibrowser properly
  • SCPP-6508 FIX: Photo (caller picture ID ) stored in internal phone book was not displayed on the phone user interface / call screen
  • SCPP-6417 FIX: The virtual keys screen was auto-closed after around a minute idle time.
  • SCPP-6294 FIX: Don't offer contrast menu for Snom MeetingPoint.
  • SCPP-6826 FIX: + sign was not accepted for a contact number via web user interface. E.g. user attempted to add a contact with +4930398330 the + was removed from the number.
  • SCPP-5938 UPD: Increase of onboard system Syslog size in web user interface: Snom 300/320/PA1: 200 entries, Snom 370: 300, all other models: 2000
  • SCPP-6821 FIX: XML definition for function key was broken. Web user interface showed its own source code due to a destroyed innerHTML.value structure, with a certain XML definition.
  • NONE FIX: Some fkey pictures on fkeys web user interface (WUI) page of snom300 were missing.
  • NONE UPD: Updated Greek language translations
  • NONE FIX: Japanese language changes
  • NONE ADD: Greek phone user interface (PUI) language added
  • SCPP-6379 FIX: XML-Browser: {index} is always 0, in SnomIPPhoneDirectory / SoftKeyItem scenarios the devices always replace the selection with 0.
  • SCPP-6282 FIX: LDAP max hits default of 50 was also the maximum value for all models, - now aligned with the documented valid value.
  • SCPP-6488 & SCPP-6618 & SCPP-6763 The LDAP search string or the current marked result was auto dialed after a few seconds (given timeout), cause the XML Dialplan was applied during LDAP search. In case an LDAP result was dialed by user before the timeout ends, the user initiated call was placed onhold by the second auto dialed attempt. Solutions using LDAP combined with an XML Dialplan containing timeout="value in seconds". Disable application of XML Dialplan during LDAP search.
  • SCPP-6761 FIX: LDAP static filter values are replaced with * . E.g. an LDAP search filter is configured with (&(memberof=a)(cn=%)) the phone sends a wrong LDAP request like: (&(memberof=*)(cn=*))
  • SCPP-6393 FIX: Malformed LDAP searchRequest with LDAP predict text enabled (T9 permutations)
  • SCPP-6551 FIX: CSTA via HTTP was broken. Sending an CSTA body to device webserver /csta interface results in a "serviceNotSupported CSTA error" response.
  • SCPP-6531 FIX: Fixed volume key event handling for plantronics bluetooth headset

Download Links

Phone ModelFile SizeSHA256 ChecksumFile Name
snom 300~3.2 MB247cced21c34b16eeaaa80568594d213
snom 320~3.2 MBb6fe752335b825483a4727654b0accba
snom 370~5.6 MB4a575b775a997e78f8b439f006d999ae
snom 370 - VPN~6.7 MB02a6f4dcfca6c12bd44db71c9daf626e
snom MP~18 MB8f8a1f7ed7a8f5fa824376e15d5a08b9