This file includes all changes made to the Convergence Line of products since their last official release being 3905.16. If you have any questions regarding this file please do not hesitate to contact support@comdasys.com. The main change in this version is support for multiport mapping in the SBC component. =================================================================================== Changes: =================================================================================== ----------------------------------------------------------------------------------- ----------------------------------------------------------------------------------- New Features: ----------------------------------------------------------------------------------- - Enhanced Servicability through new Web Interface - Enhanced service availability through watchdog application; this will restart any important service after a failure or temporary failure. It can also guard against inadvertently stopped services - Better support of TLS and TCP transport protocols - Multiport mapping feature for SBC scenarios; any client will be mapped to a unique signalling port towards the PBX - Topology hiding (eg. From, To headers) in BranchSBC and SBC - VIA Header hiding for SBC to enable a full topoglogy hiding for messages towards the Internet - Added the 'SIP TLS configuration' for VoIP to uplaod a private key and certificate or generate a default private key and certificate - Added the SSL connection check support for Siemens OptiPoint and Openstage phones ----------------------------------------------------------------------------------- Bug Fixes: ----------------------------------------------------------------------------------- - Changes to handling of 2nd PBX node in Survivability Template. The 2nd PBX node will now be handled correctly even if specified with DNS name - Bugfix to avoid changing the media connection from other end points than phones (eg. media server) cannot change media connection (noupdate, n-flag) after a ReINVITE. The problem was that the media server continued sending even after the ReINVITE and the SBC NAT handling redirected streams to the media server because it erroneously assumed this to be the media source. - Bugfixes for differing transport protocols to PBX and GWs - SIP proxy no longer runs with port 5081, fixes problems with TLS and TCP phones - Survivability user hooks now also get executed if no PBX is reachable on startup; in the past this was not the case if the box came up and stayed in Survivability mode - In dialog request routing (e.g. ReINVITE) fixed for gateway if gateway was completely masqueraded by the standard SIP port of the proxy - Do not override contact header if B2BUA is involved for gateway handling behind proxy - Fixed the problem that in rare situations multiple SIP OPTIONS packets were sent. - Fixed the problem that in rare situations the proxy tried to go into or out of survivability mode multiple times within a short time frame. - Add no rerouting response code 422 for selection in the WebGUI - Fix prefix number, and Access code digit stripping for gateway Survivability and Branch SBC templates - Fixed several bugs in the Standalone template that prevented some SIP standard features - SBC functionality enhanced with dynamic multi interface detection. The IP addresses used in SIP and SDP bodies are now only determined by the interface through which the packet gets routed. This includes dynamic interfaces such as PPPoE or VPN interfaces - Introduced TLS idle timeout which is now fixed at 10h - SIP OPTIONS ping message now contains the Siemens-X headers indicating whether the proxy is in normal or in Survivability mode - Added an extra condition for the expires header to respond with a value of 0 if a deregistration is requested - Fixed access code and digits stripping for multiple gateways for Survivability and Branch SBC - Utilize virtual IP address for Record Routes etc. if VRRD is active ----------------------------------------------------------------------------------- Additional Bug Fixes (vs 4675.17 Releases): ----------------------------------------------------------------------------------- - Fixed bug in the SSL connection check for Siemens optiPoint and Openstage phones that caused the SSL connections to break - The firewall handling was changed to properly permit TLS traffic whenever SIP TLS is used by default if Branch SBC is configured - Record Routes header in BYE messages to local gateways in the Branch SBC template contained the external instead of the internal IP address. This lead to a malfunction durign hangup if the routing to the external IP was not set - ACK with SDP processing altered in conjunction with local gateway. This caused the external IP address to be signalled to the gateway in certain scenarios leading to half way audio if routing was not set - Corrected half way audio bug if telephone outside the Branch office made a call using the locally configured gateway in the Branch Office in Branch SBC mode - Fixed problem where the SIP proxy stopped responding in some scenarios with a lot of ReINVITE messages coming from the local gateway in Survivability mode - Do not overwrite the Expires header of the Registration in normal mode and Survivability template - Hostnames of gateways in the past were not escaped correctly leading to the strange behavior that boolean terms such as "or" lead to a malfunction (SIP Proxy did not even start). This has been fixed. ----------------------------------------------------------------------------------- Additional Bug Fixes (vs 4675.18 Releases): ----------------------------------------------------------------------------------- - B2BUA now plays back correct ringback tone if a gateway call is transferred in Survivability mode with the B2BUA flag enabled - Deadlock problem fixed that could lead to the SIP Proxy not respond to SIP requests anymore; the symptom is that the SIP Proxy seems to run normally, but does not respond to any requests coming in across the network - Changes in B2BUA support to fix SIP carrier dial rules, user account dial rules as well as optimizations - Fixed SIP Proxy restart to correctly reload contacts - Fixed vrrp behaviour, openser will restart on the box that has just become active to correctly listen to the requested port - Fixed the SSL connection for Siemens endpoints - B2BUA is now properly VRRP aware and can now correctly handle the virtual IP address - Ringback tone now correctly played back in a Transfer scenario in Surv. with the B2BUA enabled. ----------------------------------------------------------------------------------- Additional Bug Fixes (vs 4675.20 Releases): ----------------------------------------------------------------------------------- - Changed timescale for TCP connect timeout from seconds to tenths of a second (default now 0.5s). This fixes a problem that if e.g. a TLS phone is unplugged and is not reachable anymore this messaging thread was blocked for several seconds. If that happened with 4 messages at the same time, all message handling threads were blocked - Do not delete /etc/sysconfig/network-scripts/.post or /etc/sysconfig/network-scripts/ifcfg-* containing the magic text DO_NOT_DELETE during update and applyconfig - Added support for second PBX node also in Branch SBC mode (ported from Survuvability) - The B2BUA now accepts SIP messages with with mixed/multipart bodies as for example used by Nortel PBXs. - Route Headers correctly removed if Topology Hiding is activated in SBC mode - The Transport Parameter is now correctly added in Record Route Headers even if Preset Record Route or VRRP virtual addresses are being used - It is now possible to correctly dial a number starting with a * in Survivability mode with B2BUA enabled. - Space optimizations were made to clear some unnecessary stuff from the flash - The B2BUA in Survivability mode now plays a correct ringback tone between initiating a Transfer and getting an Answer - No DNS Reverse Lookup queries are exectued in Survivability mode - The Cache timeout for the Survivability cache is now correctly calculated based so that the constant ping will always occur every 60 seconds and not vary based on the set timeout. - In the visualization of the SIP Proxy Registered Users, the windows is unnecessarily small - Max-Forwards header missing in ACK has been fixed - TLS - sporad. no SIP-Options to PBX sent out anymore after proxy-restart. This was a timing issue during the startup that has been resolved. ----------------------------------------------------------------------------------- Additional Bug Fixes (vs 4675.23 Releases): ----------------------------------------------------------------------------------- - Increased timeout for SIP messages handled by the SIP proxy to 200 seconds. This means that the SIP Proxy will only respond with 480 after 200 seconds and not receiving any other response to that transaction. - SIP Proxy users now again properly displayed in the WebGUI - Improvements to the WebGUI for giving proper warnings both in the SNMP page and the QoS pages - Corrected Subscribe handling in conjunction with PBX Alias Setting in Branch SBC and Survivability templates - SIP Proxy Status (Survivability or normal mode) is now properly displayed in the WebGUI again - Fixed missing Max Forwards header for ACKs sent by the Server (IP PBX) - Fixed bug in the SBC that led to improper handling of OKs with two SDP streams if the initial INVITE had no SDP at all - DHCP problem in conjunction with SIP Proxy. It is now possible for the product to get an IP address from a DHCP Server when using the SIP Proxy ----------------------------------------------------------------------------------- Additional Features (vs 4675.24 Releases): ----------------------------------------------------------------------------------- - Fix syntax errors in survivability and branch scripts for generating CDRs in Surv. mode (bug introduced only in .24 version) - Set correct interface names in bandwitdh operation instead of Lan1, Lan2 or Wan (bug introduced only in .24 version) - Fixed syntax error in Dual Registration (bug introduced only in .24 version) - Fix SIP proxy restart. Now all old registration from the dump file are restored correctly ----------------------------------------------------------------------------------- Additional Features (vs 4675.25 Releases): ----------------------------------------------------------------------------------- - Unit Manager Network Central Management Tool Support - It is possible now to do a "Factory Reset" from LCD display ----------------------------------------------------------------------------------- Changes and Bugfixes (vs 4675.26 Releases): ----------------------------------------------------------------------------------- - Fixed error in Survivability in conjunction with the Map Prefix feature that garbled the numbers - Fixed problem when directly upgrading from 3905 that Configuration file for SIP Proxy is missing because it is not properly converterd. Manual CLI intervention was necessary before - After an upgrade the SIP proxy needed to be started manually because the TLS certs were not automatically generated - B2BUA now supports correctly down negotiating from SRTP to RTP ----------------------------------------------------------------------------------- Changes and Bugfixes (vs 4675.27 Releases): ----------------------------------------------------------------------------------- - Fixed wrong record routing in conjunction with TLS and VRRP as well as the Preset Record Route Feature. In those cases the transport parameter was missing. - External Prefix Parameter was not applied correct if it contained special characters (e.g. *,#) - Fixed Record Route setting for TLS in conjunction with B2BUA mode in limited Mode - Fixed configuration generation errors in CAC handling in survivability mode - B2BUA now plays music between transfer and answer (now also works for updates and not only newly delivered boxes) - B2BUA now accepts SIPS URIs - SBC template now available on the Convergence 1600 - Fixed some issues in VIA Header Hiding for SBC - Fixed several message routing issues in conjunction with survivability and B2BUA that caused messages to be sent over UDP instead of TLS ----------------------------------------------------------------------------------- Changes and Bugfixes (vs 4675.28 Releases): ----------------------------------------------------------------------------------- - Fixes have been made to the message routing as well as the record routing in conjunction with mixed protocol operation especially if PBX speaks different protocol than the connected phones, or the phones speak different protocols - Comdasys, B2BUA in limited Mode wrong record route setting was solved - Refer instead of Forbidden is sent to gateway via Comdasys was fixed. If a phone rejected a Transfer request, the unsuccessful response was not forwarded to the gateway. Instead hunting was attempted which was of course a bug. - Fixed a memory leak in record route handling that lead to a service crash about once a week under high loads - Fixes to Apply Configuration for groups in the WebGUI ----------------------------------------------------------------------------------- Changes and Bugfixes (vs 4675.29 Releases): ----------------------------------------------------------------------------------- - Enabled support for SFTP which should solve a lot of issues with file copying with not fully compliant clients - Fixes for SURVIVABILITY NOTIFY sending to avoid phones being stuck in Surv. mode - Fixed another memory leak when relaying messages to locally connected phones (this leak was only introduced in the .27 so earlier versions did not have this problem) - Imrpoved Media Handler to give proper syslog output - SIP Proxy now inserting the X-Siemens-Proxy-State header in the registration response in Survivable mode; Siemens phones will use this to determine their status and adapt the behavior. Others will ignore this header. - Corrected Access Code Handling and Stateful routing handling with more than configured gateway - Improved handling of SIP Notify sending to avoid dropped packets - Improved handling of Session Timeouts for local and gateway calls - Fixed problem that could let to complete internal deadlock of SIP Proxy when running as an SBC - Added new CLI command force_send_notifies. This command will force a resending of the current status notification messages - Fixed bug that correctly reinitializes SIP Proxy after a VPN tunnel comes back up - Fixed a problem in SDP of OK in combination with SDP-less INVITES ----------------------------------------------------------------------------------- Changes and Bugfixes (vs 4675.30 Releases): ----------------------------------------------------------------------------------- - Fixed instabilities in conjunction with TLS and also in generic scenarios that caused occasional crashes. It is recommended to upgrade all .29 or .30 versions to this! - Please make sure to remove the ser_rtp_proxy_ip from the baseconfig. This parameter has not been configurable since 4675.9 and now must be removed for ensuring proper behavior. ----------------------------------------------------------------------------------- Changes and Bugfixes (vs 4675.31 Releases - contained in SP1): ----------------------------------------------------------------------------------- - Fixed problem in media handling that led to a signal handling process stop doing media handling after encountering one unhandled error. Now the signal handling process properly recovers and functions normal again. Only relevant for Branch SBC and SBC templates - Fixed problem with SIP devices that do not properly advertise port and transport in their contact header (only encountered with old V5 firmware of Mediatrix 1102 so far) - Watchdog is now part of the standard functionality checking the status of the SIP Proxy processes ----------------------------------------------------------------------------------- Misc Changes: ----------------------------------------------------------------------------------- - Changed the default for local gateway to "on" - Change registration timeout default to 120 in Survivability mode - Added gateway ports for configured gateways. If there is no given port, check the transmit protocol (udp, tcp or tls) and set either 5060 or 5061 as port value. - Changed the size of the tls (openssl) certificate for the https from 4096 to 1024 bits, thus accelerating the WebGUI - In Branch SBC mode a Path Header is now added to REGISTERs sent to the PBX - Added feature so that the B2BUA can take itself out of the media stream in a call ----------------------------------------------------------------------------------- Known Issues: ----------------------------------------------------------------------------------- - Call Hunting in Survivability mode is limited with TCP gateways, because the non-establishment of a PBX connection will not lead to a hunting to the next gateway. Note that this only applies to TCP and TLS connections, and only occurs if the gateway is offline for some reason. - Handling of IP addresses is not fully dynamic yet in the Branch SBC mode. Therefore it is still necessary to configure the External IP address in this mode =================================================================================== ----------------------------------------------------------------------------------- Media Gateway Component Changes (Convergence GW only), Gateway application version 1.1.11 ----------------------------------------------------------------------------------- =================================================================================== ----------------------------------------------------------------------------------- Bugs Fixes and Misc Changes: ----------------------------------------------------------------------------------- - CODEC vs. Bearer Capabilities Mapping affect SIP response. - Fax call hanging after receiving packet with media port = 0 - ISDN - Valgrind reports memory corruption and small memory leak. - The "Domain" in SIP->Gateway lacks proper description and can confuse the end user. - Applying a configuration file that contains configuration for 2 cards cause the unit to assert. - PCMU capture on T1 cards is not the proper CODEC and distortion can be heard. - The web page suggestion drop-list for the Source of a Route should not contain "hunt" - Call is lost during long vocal call - R2 : Assert when restarting R2 service during a call. - Problem to call the Head Number from the ISDN side - MOH - Wrong syslog error message is emitted when a file transfer with an empty URL is attempted. - ISDN to ISDN T1 calls do not work - Nlm - Cannot set a filter for all diagnostic traces categories. - Conf - Configuration script with alias command triggers a unit reboot. - R2 - CAS doesn't answer the I-12 signal sent by my equipment. It must complete the call or disconnect with A4. - R2 - When the emulator sends II-1, the unit sends B5. But the expected value should be B1 (Note: I changed "Line Free with Charge Tone" to B1 but the signal received is B5. If I change "Line Free without Charge Tone" to B1, the signal received is B1). - R2 - We are getting some levels near -4dBm - R2 - When unit cannot send its identification to called, it should send I-12. But the I-15 has been sent. - Call transfer doesn't work in normal and survivability mode - Unit does not fetch Configuration Script at Time of day - R2 - Unit doesn't understand the signal B-6 sent by my emulator, and send clear-forward. - SipEp : Syslog message is different in Dgw 1.1 and Dgw 2.0 - baseRateInterfaceConnectionType does not have the same default value as the Mediatrix profile - muse application crashes after invalid sdp message - R2 - Clear forward is not sent after the Congestion A-4 - Synchronize Web Page with Dgw 1.1 label 1.1.9.86 ----------------------------------------------------------------------------------- Media Gateway Component Changes (Convergence GW only), Gateway application version 1.1.13 ----------------------------------------------------------------------------------- - Re-Registration is not perform with good user name or is expected time when performing a specific scenario. - Automatically generate a MIB reference manual from the source MCD files on each build. - Nortel Certification : Add interop variables for interoperability with CS2K. - Different error message when adding invalid gateway. - SIP Registration accept invalid port number. - Half way speech path after gateways session timer expires - R2 : Channel stay in Used state when Physical Link is down - MoH is not working - Change in our Gateway page are not effective after a restart. ----------------------------------------------------------------------------------- Known Issues: ----------------------------------------------------------------------------------- - Row commands not supported by the script language. The language currently does not allow to execute a row command with the logical expected "command" syntax (for example 'Dhcp.StaticLeases[MacAddress="0090F8001234"].Delete'). However, it is possible to execute a command through a SET, by assigning the "execute" value to the cell representing the command. For example, the service DHCP's StaticLeases' Delete row command is an enum that has 2 possible values: noOp (0) and delete (10). It is possible to execute the command by assigning the value 10 to the Delete cell: Dhcp.StaticLeases[MacAddress="0090F8001234"].Delete=10 - The data codec priority set by the user may not be followed. - The Hunt Name should be less than 64 characters since "hunt-" is added when you input this hunt in a route destination. - The secondary DNS server is not tried if the primary DNS does not know the FQDN. - The new country tones such as Czech Republic have not been calibrated on the Mediatrix digital gateway. As a result, you may notice a too high output level. - Warning can be seen in fax testing equipment when performing clear channel faxes in a particular scenario. - Time criteria errors cannot be edited. You must re-enter all of the data. - Messages with 3 bytes ISDN call reference length are ignored. The maximum call reference length currently accepted is 2 bytes. If the unit receives a message with a call reference length of more than 2 bytes, it will ignore it and syslog messages "Call reference length (x bytes) is too big, ignoring" will be sent (if diagnostic traces are activated). - Inband Tone Generation does not work unless Overlap Dialing is enabled. - Some T.38 faxes fail between a Mediatrix 1402 and a Mediatrix 3000/4400. - When changing the “defaultRegistrationUnregisteredBehavior” MIB variable value, you must refresh the registration or the call may not be successful. - The contents of the persistent/Users folder is lost after a software upgrade. - The unit uses the G.726 16 kbps and G.726 24 kbps for clear-channel in some scenario. - The Mediatrix unit no longer supports the transfer version “draft-ietf-sip-cc-transfer-02.txt”. - The web page does not properly display the available functionality when the Endpoint Type is NT or TE. - The "Domain" in SIP->Gateway lacks proper description and can confuse the end user. - Problem to call the Head Number from the ISDN side.