For the first time in the history of the company, Sipwise participated in Austria´s most frequented running event, the Vienna Business Run.
The training sessions prior to the run were pulling the team closer together and were furthermore a great balance in the hectic phase before sip:provider CE v3.0 and sip:provider PRO v3.0 were released.
Congratulations to all teams for their outstanding performances, we are looking forward to the Business Run 2014!
We are excited to announce the general availability of sip:providerCE v3.1 and sip:providerPRO v3.1, which provide strong improvements of the current 3.x product line!
The Sipwise sip:provider platform is a highly versatile open source based VoIP soft-switch for ISPs and ITSPs to serve large numbers of SIP subscribers. It leverages existing building blocks like Kamailio, Sems and Asterisk to create a feature-rich and high-performance system by glueing them together in a best-practice approach and implementing missing pieces on top of it.
Sipwise engineers have been working with Asterisk and Kamailio (and its predecessors SER and OpenSER) since 2004, and have roles on the management board of Kamailio and are contributing to these projects both in terms of patches and also financially by sponsoring development tasks. The sip:provider platform is available as a Community Edition (SPCE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup.
The SPCE provides secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, file transfer, screen sharing, remote desktop control) and connect them to other SIP-, Mobile- or traditional PSTN-networks. It can therefore act as open Skype replacement system, traditional PSTN replacement, Over-The-Top (OTT) platform and also as a Session Border Controller in front of existing VoIP services in order to enable signaling encryption, IPv6 support, fraud- and Denial-of-Service prevention. Another use-case is to act as a Class4 SIP concentrator to bundle multiple SIP peerings for other VoIP services.
The majority of the changes compared to v3.0 are lots of bug fixes and usability enhancements. The most important changes are
As usual, we’re providing a VMWare Image, a Virtualbox Image and a Vagrant Box for quick evaluation testing. Check the relevant section in the Handbook for detailed instructions.
For new users, please follow the Installation Instructions in the Handbook to set up the SPCE v3.1 from scratch.
For users of the SPCE v3.0, please follow the upgrade procedure outlined in the Handbook. If you have customized your configurations using customtt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise all your calls will most certainly fail.
Over the last months we’ve started to publish our software components at github.com/sipwise. This is still an on-going effort, which is done on a component-per-component basis. Please check back regularly for new projects to appear there, and feel free to fork them and send us pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at lists.sipwise.com/listinfo/spce-dev.
We are currently working especially hard on the WebRTC part of mediaproxy-ng to implement stream multiplexing and DTLS support to keep up with the latest WebRTC changes, and we hope this will make it into v3.2.
The second important part we’re currently working on is providing you with a new REST interface, which will make it easier to develop third party applications like your own customer interfaces or integration into your soft-clients.
Also, we’re going to change our release cycle from a rough interval of ~3 months to an 8 weeks cycle to be able to bring you new features even quicker. More on the details and the support structure for new releases in a separate post.
We want to thank our PRO customers and the SPCE community for their feedback, bug reports and feature suggestions to make this release happen. We hope you enjoy using the v3.1 release and keep your input coming. A big thank you also to all the developers of Kamailio, Sems and Prosody, who make it possible for us to provide an innovative and future-proof SIP/XMPP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise development team, who worked insanely hard to create this release. You are awesome!
[Bugfix] Add pbx port to call flow graph
[Bugfix] Adopt upstream patch for milliseconds precision in acc
[Bugfix] caller avp not set when calling from foreign (Andreas Granig) – in progress.
[Bugfix] CDRs shows %23 instead of ‘#’ symbol
[Bugfix] Execute all db-schema scripts in sequential order sorted by ID revision number
[Bugfix] Fix accounting for PBX calls
[Bugfix] Fix statistics for sp-SELF
[Bugfix] In subscriber list of captured calls, show cseq_method instead of method
[Bugfix] kamailio-config-tests: review and check current tests with master
[Bugfix] kamailio-lb should mask internal PBX contact
[Bugfix] kamailio-proxy should clear $var(dpid) after use
[Bugfix] Load caller preferences of referring party in case of blind transfer
[Bugfix] Music on hold from PBX is not proxied
[Bugfix] ngcpcfg-ha: fix syntax error in functions/ha_features
[Bugfix] ngcp-installer fails on Amazon/EC2
[Bugfix] NGCP-Panel: Don’t let admin subscribers terminate their own subscriber
[Bugfix] NGCP-Panel: don’t select own contract id twice when fetching contract structure
[Bugfix] NGCP-panel: Failed to load list of resellers on WEB
[Bugfix] ngcp-panel-nginx: allow http for auto-provisioning devices
[Bugfix] NGCP-panel: Optimisation of memory usage
[Bugfix] ngcp-panel statistics don’t work if rrd name contrains non-alphanumeric chars
[Bugfix] NGCP-panel: Fix warning during WEB startup process.
[Bugfix] Nginx doesn’t start kamailio-presence server
[Bugfix] Nginx doesn’t start on PRO immediately after the installation
[Bugfix] reminder call doesn’t work for PBX users
[Bugfix] Fix weight values for Sip peering gateway
[Enhancement] Ability to debug deployment.sh (the first stage of NGCP installation)
[Enhancement] add deb-src entry (commented) into sources.list
[Enhancement] Add sems-pbx to rsyslog
[Enhancement] Allow sequential PBX device firmware upgrade
[Enhancement] collectd: Add redis plugin
[Enhancement] Don’t leak NGCP-internal headers to the UA
[Enhancement] Enable ipmi plugin in collectd
[Enhancement] Enable nginx plugin in collectd
[Enhancement] force sshd service to start early in the boot process
[Enhancement] Increase rating decimals to allow lower rates
[Enhancement] Migrate apache configs to nginx
[Enhancement] ngcpcfg: Performance optimisation of “ngcpcfg apply” (execution time)
[Enhancement] ngcp-installer fails quietly when encountering errors during ‘ngcpfg build’ run
[Enhancement] ngcp-mediaproxy-ng-daemon does not need to Pre-Depend on ngcp-mediaproxy-ng-kernel-dkms
[Enhancement] ngcp-panel: Avoid unnecessary DB lookups for ACL handling
[Enhancement] NGCP-Panel: Implement limitation for subscribers per contract
[Enhancement] ngcp-update-db-schema: avoid re-execution of ngcp-check_active
[Enhancement] no longer install kamailio + percona repository keys
[Enhancement] Only use mediaproxy for replies to INVITE
[Enhancement] Prepare sems path and config files for multiple instances
[Enhancement] proxy: route out-of-dialog Invite with Replaces to PBX
[Enhancement] Send browser to PBX device when syncing
[Enhancement] Show reseller in device management for config, firmware, profile
[New Feature] kamailio: implement hunt groups in routing
[New Feature] NGCP-Panel: Implement line-based subscriber/feature assignment for PBX product
[New Feature] ngcpsh: implement setting debug levels for sems, kam-lb and kam-proxy
When dealing with VoIP interconnections, you need to analyze the inbound and outbound call traffic from and to various countries for constantly negotiating termination fees you might receive from your peering partners, and origination fees you have to pay for.
Having an overview of your most common origination and termination countries significantly helps with figuring out where to focus on, so we started to experiment with different options how to display this information in a compact and useful way.
During our evaluation of different graph types, we came across Chord Diagrams and an example for displaying Uber Rides by Neighborhood.
Using a chord diagram, you can encode the relative frequency of a connection between two points, which in our example would be calls from one country to another and vice versa. Also, the connections are directed, so you can see immediately which of the two end points (countries) has more originations. So applying the example from above to CDRs generated by the Sipwise NGCP and integrating it into the NGCP Panel, we end up with a graph like this:
You can hover over the different country codes to highlight a certain connection, and hovering over the connections itself shows the absolute number of calls for each direction between two specific countries.
The new Call Distribution statistics will be available in the upcoming v3.2 version of the NGCP.
Easy and hassle-free NGN interconnect.
Sipwise has successfully introduced its “NGN in a box” solution providing cost effective and simple access to Deutsche Telekom’s new SIP-based NGN interconnect N-ICA.
The high degree of virtualization deployed with the Sipwise solution ensures high availability while meeting the demands of active-active SBC operations and load balancing across the NGN links. Sipwise’s NGCP based NGN SBC are tailor-made for the German interconnect landscape and allow for configuration-free no-hassle setups of N-ICA based interconnects.
With other countries expected to follow the German interconnect vanguard, Sipwise will be ready to adapt its “out-of-the-box” solution to fit other national regulatory environments.
We are excited to announce the general availability of sip:providerCE mr3.2.1 and sip:providerPRO mr3.2.1, aka the new v3.2 Version.
The Sipwise sip:provider platform is a highly versatile open source based VoIP soft-switch for ISPs and ITSPs to serve large numbers of SIP subscribers. It leverages existing building blocks like Kamailio, Sems and Asterisk to create a feature-rich and high-performance system by glueing them together in a best-practice approach and implementing missing pieces on top of it.
Sipwise engineers have been working with Asterisk and Kamailio (and its predecessors SER and OpenSER) since 2004, and have roles on the management board of Kamailio and are contributing to these projects both in terms of patches and also financially by sponsoring development tasks. The sip:provider platform is available as a Community Edition (SPCE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup.
The SPCE provides secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, file transfer, screen sharing, remote desktop control) and connect them to other SIP-, Mobile- or traditional PSTN-networks. It can therefore act as open Skype replacement system, traditional PSTN replacement, Over-The-Top (OTT) platform and also as a Session Border Controller in front of existing VoIP services in order to enable signaling encryption, IPv6 support, fraud- and Denial-of-Service prevention. Another use-case is to act as a Class4 SIP concentrator to bundle multiple SIP peerings for other VoIP services.
Previous releases were called 3.1, 3.0, 2.8 and so on. Fixes were applied to the full version, but these changes were not reflected in the name, causing two 3.1 versions to be potentially different (one without patches, one with them). The only way to force a specific version at least temporarily was to upgrade to the latest packages within the release.
Since this version 3.2, we’ve changed the versioning scheme to mr3.2.1, which is short for Maintenance Release 3.2.1. We’re going to provide regular and short-cycle upgrades (weekly or bi-weekly, depending on the number of reported issues) which will only contain bug fixes. There won’t be any new features, API changes, DB changes or whatever between say mr3.2.1 and mr3.2.2. Only really critical issues will be applied as hot-fixes into various supported versions.
Also, the current release is reflected in the Debian package version of packages provided by Sipwise. Old packages were named like ngcp-panel 1.1.10, while the new format is ngcp-panel 1.2.0.1+0~mr3.2.1.1.
This change will make it easier to distinguish between various patch levels, so if you’re going to report an issue, please provide us with the new version name you’re using, e.g. mr3.2.1.
The most important changes for mr3.2.1 are:
As usual, we’re providing a VMWare Image, a Virtualbox Image and a Vagrant Box for quick evaluation testing. Check the relevant section in the Handbook for detailed instructions.
For new users, please follow the Installation Instructions in the Handbook to set up the SPCE mr3.2.1 from scratch.
For users of the SPCE v3.1, please follow the upgrade procedure outlined in the Handbook. If you have customized your configurations using customtt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise all your calls will most certainly fail.
Over the last months we’ve started to publish our software components at github.com/sipwise. This is still an on-going effort, which is done on a component-per-component basis. Please check back regularly for new projects to appear there, and feel free to fork them and send us pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at lists.sipwise.com/listinfo/spce-dev.
In the last release notes, we mentioned a possible availability of DTLS-SRTP support for mediaproxy-ng in v3.2. The good news is that we’re proud to say that the implementation has finished and is working. The bad news is that it’s not considered stable enough to make it for mr3.2.1, but it will definitely land in mr3.3.1. In the meanwhile, you can compile the mediaproxy-ng from source if you want to try it out.
We’ve also provided a new REST interface with mr3.2.1. It’s not 100% complete yet in terms of functionality, but you can already get a good idea how it’s going to work. Most important functionality is there already (managing resellers, customers, subscribers, domains and their preferences, as well as billing profiles). For mr3.3.1 we’re going to provide the last missing pieces (refined ACLs especially for resellers, more query options for fine-grained control over filtering collections, and implementations for rewrite rules, sound sets, voicemail files etc).
Last but not least, we promised a shorter release cycle of 8 weeks. We were not able to keep up to this promise for mr3.2.1, because we had to rework a lot of our internal build and release infrastructure to support the new naming scheme, which took longer than expected. We’ve finished this tasks now, so the next release mr3.3.1 will be available in 8 weeks.
We want to thank our PRO customers and the SPCE community for their feedback, bug reports and feature suggestions to make this release happen. We hope you enjoy using the mr3.2.1 release and keep your input coming. A big thank you also to all the developers of Kamailio, Sems and Prosody, who make it possible for us to provide an innovative and future-proof SIP/XMPP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise development team, who worked insanely hard to create this release. You are awesome!
0006185: [Bugfix] lval_pvar_assign(): non existing right pvar during sending a fax
0005615: [Bugfix] API doesn’t work on Debian Wheezy due to Data::HAL issue
0005599: [Enhancement] Rework ngcp-panel auth/authz to support password-based API access
0005601: [Bugfix] editing Device Model gets removed all the keys config of PBX Device
0006047: [New Feature] Show field devices in ngcp-panel device management
0003933: [New Feature] NGCP-API: Implement subscriber handling
0003935: [New Feature] NGCP-API: Implement subscriber preference handling
0005299: [New Feature] NGCP-API: Implement billing profile handling
0005255: [New Feature] NGCP-API: Implement contact handling
0003961: [Enhancement] NGCP-Panel: Implement localization framework
0005835: [Enhancement] Translate ngcp-panel to Spanish
0005837: [Enhancement] Translate ngcp-panel to Russian
0005833: [Enhancement] Translate ngcp-panel to Italian
0004967: [Enhancement] For PBX subscribers let Extension-Subscribers “steal” alias numbers from admin subscriber
0005349: [New Feature] NGCP-API: Implement domain handling
0003931: [New Feature] NGCP-API: Implement customer contract handling
0003927: [New Feature] NGCP-API: Implement auth/authz based on client certificates
0005199: [Bugfix] API Cert needs to be stored in DB instead of file system
0003929: [New Feature] NGCP-Panel: Implement client certificate management for admins and resellers
0003949: [New Feature] NGCP-API: Implement reseller contract handling
0005215: [New Feature] NGCP-API: provide deb packages for API dependencies
0005103: [New Feature] Implement graphing call distributions
0003997: [New Feature] kamailio: implement pre-loaded route in provisioning
0006065: [Bugfix] restore $rU to the E.164 number before matching peers if force_inbound_calls_to_peer=yes
0006135: [Bugfix] snmpd doesn’t start on inactive node
0006133: [Bugfix] upgrade script must use sipwise user for mysql
0006151: [Bugfix] Contact masking is broken in 3.2 for webrtc endpoints
0006081: [Bugfix] CFT to VM has From anomymous if callee with CFT set has CLIR
0006263: [Bugfix] reject_emergency usr/dom preference doesn’t work
0006279: [Bugfix] force_outbound_calls_to_peer enabled on peer forces call to PSTN even if callee is local
0005027: [Enhancement] ngcpcfg: Performance optimisation of “ngcpcfg apply” (execution time), part #2
0005513: [Enhancement] Ability to configure tcp_children for kamailio-proxy
0005469: [Enhancement] simplify installer version handling
0005427: [Enhancement] Add -f flag to ngcp-update-db-schema to force update on active node
0006173: [Bugfix] ossbss tests were broken by commit 1055288755d03c5e7e18591aa8b58a6dc6a74bce
0005905: [Bugfix] error with apache when executing ngcpcfg apply on PRO
0002495: [New Feature] Add russian translation support
0003995: [New Feature] kamailio: implement pre-loaded route in peer routing
0006317: [Bugfix] Preferences without a label
0005325: [Enhancement] provide Debian packaging for libtcap
0005581: [Enhancement] Debianize dhtest repo
0005797: [Enhancement] Don’t put Link headers into collection GETs
0003779: [Enhancement] Some Observations on v3.0 Web Admin/Reseller from customers
0005575: [Bugfix] Clear audio cache on sems-pbx
0006209: [Bugfix] piuparts jobs failing when perl-base is involved
0005627: [Enhancement] Sems-PBX cannot relay RTP/SAVPF
0005083: [Enhancement] Port sendfax from www-csc to ngcp-panel
0006219: [Bugfix] upgrade kamailio to 4.1.2
0005571: [Bugfix] prosody ctrl port set to 0 if internal address changes
0006169: [Bugfix] Fix Debian warning obsolete-relation-form-in-source
0005967: [Bugfix] allowed_ips – add input validation
0005473: [Bugfix] ngcp-panel : Failed to create peering server. Duplicate entry “INSERT INTO `provisioning`.`voip_peer_hosts`”
0006063: [Bugfix] ngcp-panel: Cannot create device profile (DataTables warning table id = ‘configidtable’)
0006091: [Bugfix] PBX rewrite rule documentation is broken
0006023: [Bugfix] Removed glusterd restart if loadavg(5min) greater than 10 for 8 cycles of monit
0004541: [Bugfix] proxy/pbx: Implement constraints for call park/unpark
0005583: [Bugfix] user_cli is not used in case of API method send_fax
0005949: [Enhancement] serial forking based on q-value
0005929: [Bugfix] Subscriber External-ID if set is not copied to CDRs
0005995: [Bugfix] Auto upgrades 2.8->…->trunk failed (cumulative ticket for the fixes).
0005887: [Bugfix] kamailio-config-test affects ngcp-selenium-panel test
0005935: [Bugfix] firefox has been updated on selenium-client* (selenium tests stops working)
0005959: [Documentation] More generic link to Debian netinstall ISO
0003335: [Bugfix] sbc mode peer selection broken
0005921: [Bugfix] Outbound Display-Name has not double-quotes
0005645: [Enhancement] Support Romanian language in app_voicemail
0004621: [Enhancement] Add warning to log if mediaproxy is running in user space
0002563: [Bugfix] ngcp-delete_subscriber script calls non-existing get_voip_acount function
0005741: [Bugfix] NGCP-Panel doesn’t allow uppercase chars in subscriber usernames
0003553: [Bugfix] [new ngcp-panel][Android] Logout button and popup menu “Settings” doesn’t work in Chrome on Android
0001583: [Bugfix] Fix fax/voicebox/conference forwards on number change
0003745: [Enhancement] cfg-schema: provide support for config.d/$FILENAME/$REV handling
0004007: [New Feature] kamailio: implement aa/office-hours in routing
0004819: [Enhancement] Add pbx functions to XML API
0005709: [Enhancement] more flexible installer version support to easily install mr releases on demand
0005301: [Enhancement] upgrade kamailio to last stable version 4.1.0
0005587: [Enhancement] remove migration commands from apache -> nginx
0005633: [Bugfix] add kamailio presence tests
0005851: [Bugfix] kamailio-config-tests: fix lintian error
0005739: [Documentation] Document voicebox IVR hierarchy
0005225: [Bugfix] Mask/unmask local contact using the extra_socket
0005781: [Documentation] Document mobile app config
0005701: [Bugfix] strict routing support
0005771: [Bugfix] ngcp-panel: all web tests failed due to added languages support
0005677: [Bugfix] lacking conflicts in binary packages produced by template source package
0005101: [New Feature] Make force-outbound-calls-to-peer more flexible
0004003: [New Feature] kamailio: implement sla handling in routing
0005661: [Bugfix] Changing billing profile from prepaid to postpaid doesn’t delete attribute in usr_preferences
0005629: [Bugfix] External ID not saved when adding Subscriber
0005609: [Bugfix] collectd: mysql plugin: Failed to get slave statistics: `SHOW SLAVE STATUS’ did not return any rows.
0005611: [Bugfix] heartbeat: wrong ucast configuration in case if ha_iface is different on sp1 and sp2
0005613: [Documentation] Add “modarate/send” emails to spce-users@ as last step of releasing new update.
0005379: [Bugfix] kamailio-config-tests fail because of calling ROUTE_CHECK_USERPROV_CLI for null UPRN
0003713: [Bugfix] double sdp rewriting due to “udptl” hack rewriting the sdp
0005529: [Bugfix] Rate-o-mat locked on Dumper line
0005487: [Bugfix] Wrong checking for “using peer auth realm of peer”
0004009: [Enhancement] integrate kamailio tests in jenkins
0004533: [Bugfix] Implement music-on-hold in PBX
0005355: [Bugfix] Sems is not updated via XMLRPC if peer_auth_* prefs are set
0005371: [Bugfix] ngcp: helper/sync-db + sbin/ngcp-sync-constants depend on DBI module
0005343: [Bugfix] ngcp-www-cs init.d script provides wrong service name
0005321: [Bugfix] debian changelog for ngcp-sems-ha in trunk smaller then in 3.1 branch.
0005183: [Enhancement] Jenkins: create NGCP upgrade autotest
0005141: [Bugfix] PRO upgrade 3.1->trunk failed (sed: can’t read /etc/default/ipmievd: No such file or directory)
0003859: [Enhancement] provisioning and web interface logging options
We’ve seen quite a lot of interesting use cases for WebRTC, from plain P2P communication and multi-party video conferencing to server-side gesture detection etc. What doesn’t seem to be explored so much is media streaming from the server to clients, like live streaming and video-on-demand. Now media streaming is nothing new in the web (you guys heard of YouTube?), but with WebRTC you actually get a two-way media connection with both audio and video, plus your two-way signalling channel for other data. Just imagine what you could do on the server with all the audio and video coming in from your watchers.
Since we always love to experiment with cutting-edge technology at Sipwise, here is a concept for a WebRTC based media streaming platform, based purely on open source software. Since it’s using plain SIP, you can watch streams with normal SIP clients like Jitsi on your PC/Mac or CounterPath Bria on your Phone/Tablet too!
Streaming media to a browser via WebRTC requires you to deliver an audio stream encoded with Opus (or G711, which is not really a viable option due to the quality), and a video stream encoded with VP8 (or probably H264 in the future), both encrypted via DTLS-SRTP. So on one hand you need a signalling server negotiating the audio and video codecs, and on the other hand you need a media engine, transcoding the streams to the requested codecs and encrypting it with the keys, which are negotiated in-band within the media stream.
Now the idea is to be able to simply call a subscriber (e.g. stream@example.org) via SIP and get the content streamed to your client.
Since we’ve quite some tools at hand at Sipwise, the choice is fairly simple, and the architecture of our WebRTC streaming platform is going to look like this:
Let’s go over the different components and their interaction with each other:
For signalling, we’re going to use the Sipwise sip:provider CE. It’s an open source VoIP soft-switch allowing us to communicating with SIP clients over Websockets via the integrated Kamailio SIP proxy. We can also manage the users via the web interfaces and even do billing based on the duration a user is watching the stream.
We can establish media sessions both via SIP over Websockets and via normal UDP, TCP or TLS, which gives us great flexibility in hooking up different types of clients. For WebRTC in particular, we need a SIP stack in javascript, and we’re going to use tryit.jssip.net as a readily available SIP client for WebRTC.
Part of the Sipwise sip:provider CE is the rtpengine, which is a media proxy for Kamailio, developed by Sipwise. It supports transcoding DTLS-SRTP streams to normal RTP and vice versa, so we don’t need to care about the crypto part in our application server, which is going to deliver the streams.
Our application server will be the called party in the signalling stream. To make the overall architecture as simple and non-intrusive as possible for existing components, we’re just going to register it as a normal subscriber to the SIP platform. That way, you can create as many subscribers as you want with any name you want (e.g. stream@example.org, livetv@example.org, thehobbit@example.org or whatever), and register an application server for each of them, serving a particular stream or movie.
Once this subscriber is called by the SIP client of the viewer, it needs to inspect the SDP body of the INVITE message to figure out the list of supported audio and video codecs. It needs to choose a possible codec combination (we decide to support H264 and VP8 for video, and Opus and G711a for audio) and pass this information along with the media ips and ports of the caller to the media transcoder, so it can start streaming to the client.
When the call is hung up, the application server needs to tear down the streaming session and clean up after it.
For that particular building block, we decided to build our own very simple user agent in Perl based on the Net::SIP module.
The most interesting question in the planning phase was how to hook up an arbitrary media stream to our SIP session in the most simple and flexible way. Our choice was VideoLan Client (VLC), as it satisfied most of the requirements: supporting the audio and video codecs we need, be able to read all kinds of media sources, and stream them via RTP to a recipient. VLC also generates SDP for RTP streaming, but it’s not suited for SDP offer/answer, so we had to patch VLC to be able to control the RTP streams in a more fine grained way.
Using VLC, you can stream any kind of media sources, like movie files, live streams provided from somewhere else via RTSP, or even TV streams by hooking up a DVB receiver (e.g. a PCI card or even a USB stick) to your server. Yay to Live-TV over SIP!
The way we’re controlling VLC is starting it with the telnet interface, and controlling the streams via telnet commands from the application server.
Based on the architecture outlined above, we came up with a working implementation pretty quickly, and it’s working really well both on Chrome and Firefox using WebRTC:
Due to the signalling being based on SIP, it works out of the box using Jitsi too:
Although WebRTC doesn’t dictate any signaling protocol, it’s good to base the foundation on standard protocols, especially if you want to integrate your service into existing infrastructures and/or want to re-use readily available tools.
By splitting up the functional requirements into clearly defined parts, you can easily have them handled by readily available tools, and end up by only building the missing interfaces (and slight adoptions to your tools) to make them work together.
Another advantage is that you can split the functional parts physically in an easy way (as they are communicating over over the network anyways), and therefore scale it without too much pain.
Due to the two-way communication channel you’re creating with your viewers, you can get user feedback both in audio and video back to the streaming server in real-time, and possibly have chat via the signalling channel too. This opens up quite a lot of possibilities and room for creativity.
Any ideas for fun and interesting use case?
Brunn am Gebirge, 11.06.2014
Ein Großteil der Geschäftskunden vieler europäischer Netzbetreiber ist im Marktsegment der sogenannten Small Medium Enterprises (SME) angesiedelt. Bislang war eine Migration auf „All-IP“-Netze der von den Netzbetreibern angebotenen Sprachdienste in diesem Segment aus Kostengründen sehr schwierig. Durch die hohen Preise der am Markt verfügbaren ISDN/SIP-Gateways konnte kein positives Geschäftsmodell abgebildet werden. Die Folge war, dass viele Netzbetreiber ihre alte, auf „TDM“ basierende Netzinfrastruktur weiter betreiben mussten, um Sprachdienste im Marktsegment SME anbieten zu können. Der gestiegene Bedarf an schnellen Breitbandverbindungen für Geschäftskunden wird seit längerem mit sogenannten IP-basierten TK-Netzen der nächsten Generation realisiert. Somit sind viele Netzbetreiber gezwungen, weiterhin zwei Netze parallel zu betreiben, was erhebliche negative Folgen für die operativen Kosten und somit den Profit der Netzbetreiber hat.
ELCON Systemtechnik hat mit dem neuen Business Intelligent Gateway 5530 (BIG 5530) den Markt für die Migration von Sprachdiensten im Marktsegment der SME revolutioniert. Durch den sehr attraktiven Preis des BIG 5530 können Telekommunikationsanbieter endlich ein positives Geschäftsmodell realisieren, um die veraltete, auf „TDM“ basierende Technik abzuschalten und die bestehenden ISDN-Sprachdienste der Endkunden über eine neue, paketbasierte (All-IP-) Vermittlungstechnik abzubilden. Bis zu 8 ISDN-S0-Schnittstellen können auf SIP-Trunks migriert werden, wobei der Kunde weiterhin seine bestehende TK-Infrastruktur nutzen kann. Durch eine extrem genaue Taktung können „hochqualitative“ Sprach- und Fax-Dienste für Geschäftskunden umgesetzt werden. Das BIG 5530 ist sehr einfach durch einige wenige Parameteränderungen in einer Konfigurationsdatei einzustellen und kann somit in jedes Telekommunikationsnetz hinter einem WAN-Gateway beim Endkunden integriert werden. Das BIG 5530 Gateway wurde mit dem Class-5-VoIP-Softswitch von Sipwise nun ausgiebig getestet und alle bekannten ISDN-Sprachfunktionen können Endkunden problemlos zur Verfügung gestellt werden.
“Wir möchten neuen sowie bestehenden Kunden die Möglichkeit bieten, die vorhandene Infrastruktur traditioneller ISDN Telefonie in Verbindung mit moderner IP Technologie zu nützen, um von den Kostenersparnissen zu profitieren.” so Gernot Fuchs, COO bei Sipwise GmbH. „Durch die Fokussierung auf die Migration von ISDN-Sprachdiensten konnten wir ein ISDN-/ SIP-Gateway entwickeln, das durch seinen extrem günstigen Preis einen positiven Business Case für viele Netzbetreiber im Marktsegment SME ermöglicht“, sagt Werner Neubauer, CEO bei ELCON Systemtechnik.
Über E
LCON:
ELCON Systemtechnik GmbH entwickelt, produziert und liefert innovative und zukunftsorientierte Lösungen für die Telekommunikation in den Bereichen Zugangsnetze und Stromverteilung. Fokussiert auf die Optimierung von Netzen, ist ELCON Marktführer bei Netzspeise- und Netzabschlusstechnik für Privat- und Geschäftskundendienste sowie Mobilfunk-Service. Bis heute sind mehr als 10 Millionen ELCON-Produkte von den weltweit führenden Telekom-Netzbetreibern in 34 Ländern im Einsatz. ELCON wurde 1990 gegründet und hat seinen Stammsitz in Deutschland. Kern des Unternehmens sind zwei Entwicklungszentren und eine moderne Fertigungsstätte in Deutschland. Vertriebs- und Servicezentren bestehen an Standorten in Deutschland, Frankreich, Italien, Großbritannien und Österreich. Weitere Informationen sind verfügbar unter:
http://www.elcon-system.com
Ihr Ansprechpartner:
ELCON Systemtechnik GmbH
Katarina Schinke
Telefon: +49 3722 7351-331
E-Mail: katarina.schinke@elcon-system.com
Über Sipwise:
Sipwise GmbH ist in Europa der führende Anbieter für innovative Telekommunikationslösungen der nächsten Generation. Das Unternehmen bietet professionelle „Voice over IP“ und Unified Communication (UC) Lösungen auf Hardware- und Softwarebasis an. Neben namhaften Telekom-Anbietern wie Liberty Global UPC gewinnt Sipwise immer mehr große, mittlere und kleinere Netzbetreiber als Kunden. Zuletzt kamen auf dem deutschen und dem Schweizer Markt etliche renommierte Unternehmen als Neukunden hinzu.
Sipwise wurde 2008 gegründet und hat seinen Firmensitz in Brunn am Gebirge, Österreich. Mehr dazu unter www.sipwise.com
Ihr Ansprechpartner:
Sipwise GmbH, Europaring F15, 2345 Brunn am Gebirge, Österreich
Susanne Windisch
Telefon: +43(0)130120-12
E-Mail: swindisch@sipwise.com
Brunn am Gebirge, June 18, 2014
Sipwise is making a decisive step forward with its new products at this year’s ANGACOM, presenting the Cloud PBX, the Unified Communication Server and the new Smartphone App.
“Broadband meets content” was the organising motto behind the internationally leading Business Platform ANGA COM, which took place this year between May 20 and 22 in Cologne. More than 450 represented firms provided some 17,000 visitors with a glimpse of new trends in the networking industry. Innovative solutions addressing the themes of communication and interaction were clearly central, along with new and highly interesting suggestions for combining old and new technology.
Several exhibitors have identified this need and offered possible solutions to the challenge of providing Hard- and Software solutions for various requirements. Sipwise has especially set the tone in this respect.
The simple solution is equally flexible in its setup and designed for all kinds of user specifications.
“This year’s ANGA COM demonstrated again that Sipwise´s products hit the nerve of future demands!” Andreas Granig, Managing Director of R&D at Sipwise proudly recognizes, feeling reconfirmed after three successful and exhausting days in Cologne.
About Sipwise
Sipwise is the leading provider of innovative telecommunication solution of the next Generation. The company offers professional „Voice over IP“ and Unified Communication (UC) solutions on hard- and software basis. Lately Sipwise introduced successfully its “NGN in a box” solution providing cost effective and simple access to Deutsche Telekom’s new SIP-based NGN interconnect N-ICA.
Sipwise was founded in 2008 in Austria and operates globally with currently twenty employees. Development, consulting and sales are covered under one roof.
For questions and further information, please contact:
Sipwise GmbH
Susanne Windisch
Europaring F15, 2345 Brunn am Gebirge
Email: swindisch@sipwise.com, Office: +43(0)130120-12
Internet: www.sipwise.com
We are excited to announce the general availability of sip:providerCE mr3.3.1 and sip:providerPRO mr3.3.1, aka the new v3.3 Version.
The Sipwise sip:provider platform is a highly versatile open source based VoIP soft-switch for ISPs and ITSPs to serve large numbers of SIP subscribers. It leverages existing building blocks like Kamailio, Sems and Asterisk to create a feature-rich and high-performance system by glueing them together in a best-practice approach and implementing missing pieces on top of it.
Sipwise engineers have been working with Asterisk and Kamailio (and its predecessors SER and OpenSER) since 2004, and have roles on the management board of Kamailio and are contributing to these projects both in terms of patches and also financially by sponsoring development tasks. The sip:provider platform is available as a Community Edition (SPCE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup.
The SPCE provides secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, file transfer, screen sharing, remote desktop control) and connect them to other SIP-, Mobile- or traditional PSTN-networks. It can therefore act as open Skype replacement system, traditional PSTN replacement, Over-The-Top (OTT) platform and also as a Session Border Controller in front of existing VoIP services in order to enable signaling encryption, IPv6 support, fraud- and Denial-of-Service prevention. Another use-case is to act as a Class4 SIP concentrator to bundle multiple SIP peerings for other VoIP services.
The most important changes for mr3.3.1 are:
As usual, we’re providing a VMWare Image, a Virtualbox Image and a Vagrant Box for quick evaluation testing. Check the relevant section in the Handbook for detailed instructions.
For new users, please follow the Installation Instructions in the Handbook to set up the SPCE mr3.3.1 from scratch.
For users of the SPCE mr3.2.x, please follow the upgrade procedure outlined in the Handbook. If you have customized your configurations using customtt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise all your calls will most certainly fail.
Over the last months we’ve started to publish our software components at github.com/sipwise. This is still an on-going effort, which is done on a component-per-component basis. Please check back regularly for new projects to appear there, and feel free to fork them and send us pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at lists.sipwise.com/listinfo/spce-dev.
The mr3.3.1 release contains a huge number of new features considering the short release cycle. For the upcoming mr3.4.1 we’re preparing the system for a new LTS version, so it will mainly be optimization and bug fixing. Nonetheless, we’re currently working on Event Detail Records (compared to Call Detail Records), so you could charge customers based on the features they are using instead of the calls, allowing for attractive product modeling. Also, there will be some further enhancements to the REST-API, so any feedback on the current state is highly appreciated, as it is the first version we consider stable and feature-complete. The SOAP/XMLRPC API does not receive any new features anymore, so this is a good time to consider switching to the new API.
We want to thank our PRO customers and the SPCE community for their feedback, bug reports and feature suggestions to make this release happen. We hope you enjoy using the mr3.3.1 release and keep your input coming. A big thank you also to all the developers of Kamailio, Sems and Prosody, who make it possible for us to provide an innovative and future-proof SIP/XMPP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise development team, who worked insanely hard to create this release. You are awesome!
MT#7569 fix CLIR for PBX subscribers
MT#7535 Remove REFER parameter from Allow header in case of Allow_refer is NO
MT#7531 kamailio-config-tests add label ERROR to tap output to find broken place quickly
MT#7501 Change order of services startup on failover
MT#7499 add config.yml option to disable 100rel
MT#7495 Implement contract preference ncos and block handling
MT#7487 NGCP-API: allow filter on profile_id in subscribers
MT#7485 NGCP-API: allow filter on soundset name
MT#7471 Introduce contract preferences
MT#7469 ngcp-panel: clean-up device provisioning config
MT#7455 P-* headers are duplicated in ROUTE_EXECUTE_CF_LOOP on hunting
MT#7453 CloudPBX: implement Cisco SPA directory service
MT#7449 CloudPBX: provide config template for spa-1xx and spa-2xx
MT#7439 ngcp-panel: Fix typo in the Call Through label
MT#7421 rest api: empty filename of soundfile returns error
MT#7417 monit: heartbeat failover lasts more than 30 seconds
MT#7415 ngcp-panel: delete ncos assignments when deleting an ncos level
MT#7407 Move sems option media_processor_threads to config.yml
MT#7405 NGCP-API: allow filter on folders for /api/voicemails
MT#7399 create sems cache directory
MT#7397 create subscribers through SOAP interface fail.
MT#7395 API REST: requests api for managing NCOS levels
MT#7377 CloudPBX: add dedicated is_pbx_pilot field to subscribers instead of using admin flag
MT#7355 sems restart sometimes exit with non-zero exit code (missed –oknodo)
MT#7347 sems crashes in dsm module on sems stop
MT#7323 REST API: desire API support a method to retrieve all handles of a collection
MT#7309 REST API: unable to create/update autoattendant slots
MT#7307 REST API: unpossible to reset speeddial
MT#7275 templates/lsb_scripts are out of sync with debian/${init} scripts
MT#7271 REST API: Callforward settings are set, but not applied.
MT#7265 Fix back button for reseller in panel
MT#7239 REST API: clir can’t be set to false
MT#7233 Implement basic number block management (WIP)
MT#7219 NGCP-API: document HTTP error codes
MT#7211 API: implement autoattendant
MT#7199 Add iban and bic to contacts
MT#7191 ngcp-panel: select country in contact by datatables
MT#7183 Changing GUI language subscribers/customers are terminated without confirmation windows
MT#7179 aa_welcome sound is not deleted from cache when a new file is uploaded
MT#7175 NGCP-API: make role handling more robust in api doc
MT#7171 ngcp-panel-rest-api broken by huge OPTION reply on URI :1443/API/
MT#7169 max_concurrent_calls should exclude pbx-internal calls
MT#7155 ngcp-panel init script can be started in parallel, leaving untraced processes behind
MT#7153 Missing Create PBX Group button if not defined the max extension number in the customer
MT#7141 upgrade packaging style to 3.0
MT#7137 deployment.sh doesn’t rebuild configs for adjust_for_low_performance mode
MT#7131 sipwise-kamailio-config-tests produce warning on start
MT#7119 REST API: add order_by as query filter for GET API calls
MT#7113 REST API: missing possibility to define/update hunt-policy and hunt-timeout in subscriber
MT#7109 kamailio-config-tests: add kamailio mem info
MT#7099 update sems-pbx core to latest upstream
MT#7095 Document pbx device bootstrap, monitoring oids and device profiles
MT#7089 Fix url-encoded ; in Route of CANCEL and ACK
MT#7077 CloudPBX: Fix SPA handling with dot in SIP username
MT#7067 NGCP-API: Implement sound handling
MT#7065 fixing some lintian errors
MT#7061 NGCP-API: Implement reminder settings
MT#7053 mod_sipwise_vjud can’t connect to mysql
MT#7051 Webusername – if not set – is not initialize using API
MT#7039 NGCP-API: Implement SIP registration handling
MT#7037 CloudPBX: Linksys SPA configuation tweaks
MT#7029 REST API: Can’t create subscriber with specific customer_id
MT#7007 Use dom/usr outbound_* preferences when force_outbound_calls_to_peer=1
MT#7005 wrong call flow for PBX users with CFU
MT#7003 set ringtimeout only once in case callee is a PBX user with CFT
MT#6985 REST API: filter subscribers by customer_id field
MT#6981 kamailio: add variables support to dialplan module rules
MT#6979 REST API: include id property in pbxdeviceprofiles response
MT#6969 ngcp-panel: trigger SPA device resync via SIP NOTIFY
MT#6967 ngcp-panel: implement number range handling for aliases and cleanup subscriber editing as a whole
MT#6963 Remove wheezy-backports repository from NGCP installations
MT#6959 Replace huge Conflicts sections to Provides for ngcp-upgrdae
MT#6931 Make voicebox dial-in match more tight
MT#6919 mysql prosody user has allways the same password in ALL installations
MT#6913 NGCP-API: Manage profiles and profile sets
MT#6909 NGCP-API: implement email templates
MT#6907 mysql upgrade 5.5.35->5.5.37changes the password and mysql init script stops working
MT#6895 ngcpcfg: Improve the output/error reporting for no-root users
MT#6871 Updating subscribers using REST fails
MT#6865 SLA feature: sems-pbx crashes if the immediate NOTIFY times out
MT#6863 Hide webfax link if faxserver is not enabled
MT#6849 ngcp-panel: doesn’t start in trunk, lacking libcatalyst-plugin-email-perl
MT#6843 Implement automatic password generation for subscribers
MT#6841 Customer section in Create Sound Set menu gets mixed up
MT#6839 ngcp-panel: fix lintian errors
MT#6835 Set UPRN in case of call deflection (302 message)
MT#6833 ngcp-panel: uses the /etc/ssl/ngcp/api directory but we are using /etc/ngcp-panel/api_ssl/ now
MT#6831 Implement global password policy
MT#6829 ngcp-upgrade: line 169: etckeeper: command not found
MT#6827 NGCP-API: Implement pbxdevices for cloudpbx customers
MT#6819 support squeeze-lts repository in our LTS release
MT#6817 failing kamailio-config-tests in mr3.2.1 PRO
MT#6815 ngcp-panel/ngcp-templates-*ngcp-panel installation broken in trunk
MT#6811 CFT doesn’t work for PBX users
MT#6807 systemd support for our stack
MT#6789 Email notification after subscriber creation and for password reset
MT#6783 block_out_override_pin doesn’t work after pbx loop
MT#6775 sems-ha VSC module doesn’t know about $avp(s:caller_cloud_pbx_base_cli)
MT#6773 subscsriber preferences API call doesn’t return the collection reopened
MT#6771 http_proxy prevents applieing of voisniff configuration
MT#6761 Glusterfs migration script fails when proxy is configured
MT#6731 ngcp-panel: Failed to create reseller (no such column)
MT#6729 Cloud PBX: skip CF to local endpoints when hunting
MT#6727 XMLRPC function call fails if the response include custom values
MT#6695 Separate certificate for REST API
MT#6693 ngcp-panel: implement subscriber profiles for subscriber csc feature control
MT#6687 ngcp-panel: use db transaction when cloning rewrite rule set
MT#6683 Possibility to enable/disable SIP INFO method
MT#6667 Update documentation to use user ‘root’ instead of ‘sipwise’ (which has no password access anymore)
MT#6665 False Billing Fee successfully uploaded message
MT#6663 Contract (billing.billing_mappings) should always have billing_profile
MT#6659 Prepare and release mr3.3.1 (release 3.3 build 1)
MT#6657 Prepare and release mr3.2.2 (release 3.2 build 2)
MT#6633 Fix stdout redirection in rate-o-mat daemonize procedure
MT#6611 ngcp-rtpengine-daemon fails to start on current trunk installations
MT#6565 kamailio-config-tests: fix problems on heavy load systems
MT#6563 mr3.2.1->trunk upgrade is broken (ngcp-services-pro, ngcp-templates-pro have been kept back)
MT#6561 ngcp-ngcp-pro cannot be removed
MT#6559 reduce perl dependencies
MT#6551 ngcp-panel: Improve Build script
MT#6543 Generate TAP results for kamailio-config-tests if no sources were found
MT#6525 ngcp-sems-ha depends on external-package sems-python-modules
MT#6523 lua-lemock depends on binary version of lua-unit package
MT#6521 ngcp-templates-pro-hylafaxplus fails to install in clean environment due to dependency on ngcpcfg
MT#6517 unify Debian packaging
MT#6513 ngcp-panel: usage of Replaces/Breaks in Debian package
MT#6511 Bump Standards-Version of Debian packages to current policy version
MT#6509 Prosody doesn’t start due to missed port (redis shared location storage regression)
MT#6507 Ship bootlogd by default
MT#6501 Install ngcp-templates-pro-hylafaxplus-iax by default on PRO
MT#6499 kamailio-config-tests: fix problems on EC2 environment
MT#6497 API: Implement callforward handling for subscribers
MT#6495 acc-cleanup tool fails to DUMP cdr table cause /var/backups/cdr doesn’t exist
MT#6493 Time set shows obsolete years
MT#6491 Calling route(ROUTE_CLIR) from route[ROUTE_FIND_CALLEE] instead of route[ROUTE_INVITE]
MT#6487 ngcp-panel: support self-signup
MT#6479 NGCP-API: Review admin/reseller ACL
MT#6469 Prosody: Loaded modules
MT#6463 Server string still says NGCP 2.X
MT#6461 Load reseller-specific css on top of system css
MT#6459 Update peering contract and peering auth prefs fails
MT#6457 Add rtpengine support to monit (‘mediaproxy’ failed to start)
MT#6443 Dashboard calculates Customer revenue wrong
MT#6425 Integrate translation check into nightly builds
MT#6423 error upgrading to the latest version of ngcp-mediaproxy-ng-daemon
MT#6411 prosody: shared location storage
MT#6409 prosody: be able to connect to admin_telnet from the internal network
MT#6405 kamailio/lb: implement new option dos_whitelisted_subnets
MT#6389 Fix system-tester and integrate it into daily-build-testrunner
MT#6363 piuparts error: glusterfs-client: depends-on-obsolete-package depends: fuse-utils (>= 2.7.4)
MT#6359 Inconsistent kamailio lcr_gw, lcr_rule_target handling for gateways with non-unique IP
MT#6345 move libtcap module to its own debian package
MT#6335 kamailio-config-tests: invite_allowip should check for ANY integer allowed_ips_grp
MT#6323 Mysql plug-in of Collectd fail to connect if root password set
MT#6317 Preferences without a label
MT#6295 fix api tests: preference no longer required
MT#6285 prosody 0.9.3 released
MT#6283 NGCP-API: Implement search framework for collections
MT#6279 force_outbound_calls_to_peer enabled on peer forces call to PSTN even if callee is local
MT#6277 installer: support VPC with elastic IP on EC2
MT#6271 Change contact postcode field type from int to string in SOAP Provisioning
MT#6265 Interruptions not managed correctly
MT#6263 reject_emergency usr/dom preference doesn’t work
MT#6243 allow multipart body on kamailio
MT#6219 upgrade kamailio to 4.1.2
MT#6207 API get_sip_peering_contracts replies with ValueError: invalid literal for int()
MT#6195 NGCP-API: Implement rewrite rule handling
MT#6191 Intermediate acc records (acc failsafe)
MT#6185 lval_pvar_assign(): non existing right pvar during sending a fax
MT#6173 ossbss tests were broken by commit 1055288755d03c5e7e18591aa8b58a6dc6a74bce
MT#6169 Fix Debian warning obsolete-relation-form-in-source
MT#6165 kamailio-config-tests: reduce running time
MT#6151 Contact masking is broken in 3.2 for webrtc endpoints
MT#6141 Impossible to switch active node to sp2 (hb_watchdog returns sctive status tp sp1)
MT#6135 snmpd doesn’t start on inactive node
MT#6133 upgrade script must use sipwise user for mysql
MT#6119 SOAP function set_subscriber_preferences configure wrong ‘allowed_ips’ references if network is the same
MT#6101 provide libtcap for squeeze + 2.8 release
MT#6091 PBX rewrite rule documentation is broken
MT#6081 CFT to VM has From anomymous if callee with CFT set has CLIR
MT#6075 stop monit on upgrade script
MT#6065 restore $rU to the E.164 number before matching peers if force_inbound_calls_to_peer=yes
MT#6039 Implement ECR
MT#6025 templates-piuparts fails because of problems installing diva-drivers
MT#6023 Removed glusterd restart if loadavg(5min) greater than 10 for 8 cycles of monit
MT#6021 Parallel forking with 302 – per branch failure routing block
MT#6013 Race condition in contract balances creation
MT#5959 More generic link to Debian netinstall ISO
MT#5949 serial forking based on q-value
MT#5947 kamailio-config-tests: add ban/dos tests scenarios
MT#5933 Deleting one trusted source from GUI cause delettion of all entries with same IP in DB
MT#5921 Outbound Display-Name has not double-quotes
MT#5911 Provide single entry-point for websocket connections
MT#5905 error with apache when executing ngcpcfg apply on PRO
MT#5891 clean lintian warnings from kamailio
MT#5889 Cleanup warnings from sems-*-binaries jenkins jobs
MT#5887 kamailio-config-test affects ngcp-selenium-panel test
MT#5879 PDF Invoice generation
MT#5865 kamailio-config-tests: Add more documentation
MT#5851 kamailio-config-tests: fix lintian error
MT#5837 Translate ngcp-panel to Russian
MT#5811 Add UC client to ngcp-panel reopened
MT#5789 kamailio-config-tests: add call from/to foreign domain
MT#5787 kamailio-config-tests: add max concurrent calls tests
MT#5775 ringtimeout preference can’t be changed via SOAP
MT#5751 avoid sleep X statements in selenium tests
MT#5715 heartbeat: sometimes cannot be restarted due to error in log
MT#5701 strict routing support
MT#5645 Support Romanian language in app_voicemail won’t fix
MT#5599 Rework ngcp-panel auth/authz to support password-based API access
MT#5583 user_cli is not used in case of API method send_fax
MT#5571 prosody ctrl port set to 0 if internal address changes
MT#5473 ngcp-panel : Failed to create peering server. Duplicate entry INSERT INTO `provisioning`.`voip_peer_hosts`
MT#5349 NGCP-API: Implement domain handling
MT#5301 upgrade kamailio to last stable version 4.1.0
MT#5251 Significantly improved fees upload speed (Fixed 504 Gateway Time-out during upload fees)
MT#5103 Implement graphing call distributions
MT#5101 Make force-outbound-calls-to-peer more flexible
MT#5083 Port sendfax from www-csc to ngcp-panel
MT#4967 For PBX subscribers let Extension-Subscribers steal alias numbers from admin subscriber
MT#4921 Enhance PBX installation process
MT#4865 enable SRTP if there is a m-line with RTP/SAVP profile
MT#4825 heartbeat: hb_watchdog is not running by default on PRO
MT#4759 Lb sends to proxy always messages with Max-forwards header set to 16
MT#4541 proxy/pbx: Implement constraints for call park/unpark
MT#4489 Add log information to mitigate brute force attacks
MT#4393 Call forwarding loops create high system load
MT#4369 Sip Calls Flow page seems is not working
MT#4363 Fixed init daemon dependency (proper starting order)
MT#4235 Migrate JitsiProvisioning from www-csc to ngcp-panel
MT#3955 NGCP-API: Implement fetching of call costs for given billing interval
MT#3947 NGCP-API: Implement voicemail settings handling for subscribers
MT#3945 NGCP-API: Implement faxserver settings handling for subscribers
MT#3943 NGCP-API: Implement call list handling for subscribers
MT#3941 NGCP-API: Implement voicemail handling for subscribers
MT#3939 NGCP-API: Implement trusted source handling for subscribers
MT#3937 NGCP-API: Implement speed-dial handling for subscribers
MT#3935 NGCP-API: Implement subscriber preference handling
MT#3933 NGCP-API: Implement subscriber handling
MT#3931 NGCP-API: Implement customer contract handling
MT#3335 Fix sbc mode peer selection
We are excited to announce the general availability of sip:providerCE mr3.3.2 and sip:providerPRO mr3.3.2, aka the next build of release v3.3 Version.
The Sipwise sip:provider platform is a highly versatile open source based VoIP soft-switch for ISPs and ITSPs to serve large numbers of SIP subscribers. It leverages existing building blocks like Kamailio, Sems and Asterisk to create a feature-rich and high-performance system by glueing them together in a best-practice approach and implementing missing pieces on top of it.
Sipwise engineers have been working with Asterisk and Kamailio (and its predecessors SER and OpenSER) since 2004, and have roles on the management board of Kamailio and are contributing to these projects both in terms of patches and also financially by sponsoring development tasks. The sip:provider platform is available as a Community Edition (SPCE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup.
The SPCE provides secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, file transfer, screen sharing, remote desktop control) and connect them to other SIP-, Mobile- or traditional PSTN-networks. It can therefore act as open Skype replacement system, traditional PSTN replacement, Over-The-Top (OTT) platform and also as a Session Border Controller in front of existing VoIP services in order to enable signaling encryption, IPv6 support, fraud- and Denial-of-Service prevention. Another use-case is to act as a Class4 SIP concentrator to bundle multiple SIP peerings for other VoIP services.
The build mr3.3.2 contains the set of fixes for release mr3.3. No new features added to mr3.3.2. See the full list of changes at the end of the announce.
As usual, we’re providing a VMWare Image, a Virtualbox Image and a Vagrant Box for quick evaluation testing. Check the relevant section in the Handbook for detailed instructions.
For new users, please follow the Installation Instructions in the Handbook to set up the SPCE mr3.3.2 from scratch.
For users of the SPCE mr3.2.x, please follow the upgrade procedure outlined in the Handbook. If you have customized your configurations using customtt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise all your calls will most certainly fail.
Over the last months we’ve started to publish our software components at github.com/sipwise. This is still an on-going effort, which is done on a component-per-component basis. Please check back regularly for new projects to appear there, and feel free to fork them and send us pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at lists.sipwise.com/listinfo/spce-dev.
The mr3.3.2 build is stabilization/bugfixes build, so contains no new features. For the upcoming mr3.4.1 we’re preparing the system for a new LTS version, so it will mainly be optimization and bug fixing. Nonetheless, we’re currently working on Event Detail Records (compared to Call Detail Records), so you could charge customers based on the features they are using instead of the calls, allowing for attractive product modeling. Also, there will be some further enhancements to the REST-API, so any feedback on the current state is highly appreciated, as it is the first version we consider stable and feature-complete. The SOAP/XMLRPC API does not receive any new features anymore, so this is a good time to consider switching to the new API.
We want to thank our PRO customers and the SPCE community for their feedback, bug reports and feature suggestions to make this release happen. We hope you enjoy using the mr3.3.2 build and keep your input coming. A big thank you also to all the developers of Kamailio, Sems and Prosody, who make it possible for us to provide an innovative and future-proof SIP/XMPP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise development team, who worked insanely hard to create this release. You are awesome!
We are excited to announce the general availability of sip:providerCE mr3.4.1 and sip:providerPRO mr3.4.1, aka the new v3.4 Version.
The Sipwise sip:provider platform is a highly versatile open source based VoIP soft-switch for ISPs and ITSPs to serve large numbers of SIP subscribers. It leverages existing building blocks like Kamailio, Sems and Asterisk to create a feature-rich and high-performance system by glueing them together in a best-practice approach and implementing missing pieces on top of it.
Sipwise engineers have been working with Asterisk and Kamailio (and its predecessors SER and OpenSER) since 2004, and have roles on the management board of Kamailio and are contributing to these projects both in terms of patches and also financially by sponsoring development tasks. The sip:provider platform is available as a Community Edition (SPCE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup.
The SPCE provides secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, file transfer, screen sharing, remote desktop control) and connect them to other SIP-, Mobile- or traditional PSTN-networks. It can therefore act as open Skype replacement system, traditional PSTN replacement, Over-The-Top (OTT) platform and also as a Session Border Controller in front of existing VoIP services in order to enable signaling encryption, IPv6 support, fraud- and Denial-of-Service prevention. Another use-case is to act as a Class4 SIP concentrator to bundle multiple SIP peerings for other VoIP services.
The most important changes for mr3.4.1 are:
As usual, we’re providing a VMWare Image, a Virtualbox Image and a Vagrant Box for quick evaluation testing. Check the relevant section in the Handbook for detailed instructions. Also you can use our AMI (Amazon Machine Images) image or Docker container.
For new users, please follow the Installation Instructions in the Handbook to set up the SPCE mr3.4.1 from scratch.
For users of the SPCE mr3.3.x, please follow the upgrade procedure outlined in the Handbook. If you have customized your configurations using customtt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise all your calls will most certainly fail.
Over the last months we’ve started to publish our software components at github.com/sipwise. This is still an on-going effort, which is done on a component-per-component basis. Please check back regularly for new projects to appear there, and feel free to fork them and send us pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at lists.sipwise.com/listinfo/spce-dev.
We want to thank our CARRIER and PRO customers and the SPCE community for their feedback, bug reports and feature suggestions to make this release happen. We hope you enjoy using the mr3.4.1 release and keep your input coming. A big thank you also to all the developers of Kamailio, Sems and Prosody, who make it possible for us to provide an innovative and future-proof SIP/XMPP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise development team, who worked insanely hard to create this release. You are awesome!
MT#8367 Rest api: fixed Mysql error on reseller login
MT#8287 Failed to create PBX extension (pilot subscriber can’t create extensions)
MT#8261 REST API: api/cftimesets unable to create time set with minutes interval starting with ’0′
MT#8259 Rtpengine session is not destroyed in ROUTE_STOP_RTPPROXY_BRANCH
MT#8247 Search sipstats by UUID fails
MT#8171 Asterisk/ngcpcfg.services doesn’t check active node
MT#8147 Prepare and release mr3.3.2 (release 3.3 build 2)
MT#8121 Incoming calls interrupted afer 90 upto 180 seconds (fix adding P-Out-Socket to reply, remove ambigous checks)
MT#8093 Fix instructions for downloading ngcp-installer in 3.x handbooks
MT#8065 Problem with billing profile changing
MT#7995 Using E.164 internally instead of local numbering
MT#7985 Default ntp.conf does not work no change required
MT#7981 Changing admin flag for subscribers not possible in admin panel
MT#7959 REST API: contracts creation require property type
MT#7927 Cannot SSH system using cdrexport account (su: Cannot determine your user name.)
MT#7901 Upgrade 3.1=>mr3.2.2: strange behavior at the end of upgrade process
MT#7867 Logo inside PDF invoice can’t be changed reopened
MT#7861 Invoice template creation issue
MT#7843 VAT Rate and VAT included missing from Billing Fees
MT#7841 API: Improve documentation on callforwards
MT#7803 Show invoice links and sections in panel
MT#7797 API: allow to filter for customer status
MT#7785 Rewrite rules are not engaged for 302 redirect from pbx users
MT#7783 Proxy to support multi-value Privacy header
MT#7737 Action methods (begin) found defined in your application class
MT#7731 Can’t see/create soundset as a reseller
MT#7717 Domain are present in DB but not in GUI reopened
MT#7715 Failover problem on calls from PBX to PSTN
MT#7713 Auto Attendand Slot menu is available for Standars SIP account (not pbx) user preferences
MT#7685 Provisioning.wsdl – add ‘nilable’ flag into SubscriberPreferencesRead
MT#7683 SOAP – broken data types
MT#7655 No Edit button on Cloud PBX subscribers preferences mr3.2.2
MT#7647 Hylafax templates – minor fixes
MT#7645 Diva-drivers adaptation for kernel 3.13
MT#7631 Hb_watchdog switch back active node if it checks in the same second of manual switching.
MT#7595 Reloading sound file doesn’t work – move sw_audio_api.dsm from callingcard to templates package
MT#7585 documentation about administrator user/passwd + panel
MT#7573 Upgrade script should cleanup old core files
MT#7569 Per-call CLIR is broken for PBX subscribers
MT#7557 REST API: unable to change pbx extension of a subsrciber
MT#7535 Remove REFER parameter from Allow header in case of Allow_refer is NO
MT#7531 Kamailio-config-tests add label ERROR to tap output to find broken place quickly
MT#7501 Change order of services startup on failover
MT#7499 Add config.yml option to disable 100rel
MT#7495 Implement contract preference ncos and block handling
MT#7493 NGCP Panel do not show Auto Attendat detination as option for Call Forwarding no change required
MT#7471 Introduce contract preferences
MT#7469 Ngcp-panel: clean-up device provisioning config
MT#7455 P-* headers are duplicated in ROUTE_EXECUTE_CF_LOOP on hunting
MT#7447 Ngcp-panel-rest-api broken in trunk (mr3.4)
MT#7417 Monit: heartbeat failover lasts more than 30 seconds
MT#7407 Move sems option media_processor_threads to config.yml
MT#7399 Create sems cache directory
MT#7397 Create subscribers through SOAP interface fail.
MT#7377 CloudPBX: add dedicated is_pbx_pilot field to subscribers instead of using admin flag
MT#7355 [PRO only] sems restart sometimes exit with non-zero exit code (missed –oknodo)
MT#7351 Hylafax – force set enveloper-from
MT#7347 Sems crashes in dsm module on sems stop
MT#7275 Templates/lsb_scripts are out of sync with debian/${init} scripts
MT#7239 REST API: clir can’t be set to false
MT#7199 Add iban and bic to contacts
MT#7155 Ngcp-panel init script can be started in parallel, leaving untraced processes behind
MT#7141 Upgrade Debian packaging style to 3.0
MT#7137 Deployment.sh doesn’t rebuild configs for adjust_for_low_performance mode
MT#7005 Wrong call flow for PBX users with CFU
MT#6963 Remove wheezy-backports repository from NGCP installations
MT#6705 Make clir preference more flexible
MT#6693 ngcp-panel: implement subscriber profiles for subscriber csc feature control
MT#6135 Anmpd doesn’t start on inactive node
MT#5879 PDF Invoice generation improvements
MT#5789 Kamailio-config-tests: add call from/to foreign domain
MT#5775 Ringtimeout preference can’t be changed via SOAP
Der österreichische Technologielieferant Sipwise überzeugt NetCologne mit innovativen Produkten und professioneller Dienstleistung.
Köln / Brunn am Gebirge im August 2014
NetCologne bietet als regionaler Netzbetreiber im Großraum Köln/Bonn/Aachen seit 20 Jahren Privat- und Geschäftskunden zukunftssichere Kommunikationstechnologie und -leistungen. Hohe Qualitätsanforderungen und der sich rasant ändernde Markt erfordern laufend Investitionen und Erneuerung der IT Infrastruktur. Die Entscheidung für Sipwise und ihr erfolgreiches Produktportfolio im Kommunikationsbereich trägt diesem Konzept Rechnung. Frank Ewen, Leiter Switching und Voice bei NetCologne, fasst zusammen: „Die Entscheidung fiel auf Sipwise, weil uns der Open-Source-Ansatz und die vielen Features von Sipwise überzeugt haben. Damit lassen sich Aufwände reduzieren und gleichzeitig ein hochwertiges Sprachnetz betreiben. In der aktuellen Implementierungsphase zeigt sich Sipwise als professioneller und zuverlässiger Partner.“
Als Anforderung für Sipwise als Technologielieferant und Servicepartner galt, dem Wunsch nach einer multifunktional nutzbaren Lösung zu entsprechen, um höchstmögliche Flexibilität auch in Zukunft zu gewährleisten. Die neue Plattform sollte ein möglichst breites Spektrum abdecken und die idealen Voraussetzungen sowohl für Privat- als auch für Geschäftskunden bieten. „Ausschlaggebend für die Wahl der Lösung von Sipwise ist neben der umfassenden Funktionalität und Flexibilität besonders die einfache und schnelle Integrationsmöglichkeit in unsere bestehende Systemumgebung. Die Herausforderung besteht darin, dass die Betriebssicherheit unseres gesamten Netzwerks trotz eines engen Zeitplans in jeder Phase sichergestellt sein muss. Bislang arbeitet das Projektteam hier hervorragend.“ erläutert Frank Ewen.
„NetCologne kann mit Hilfe der fortschrittlichen Technologie von Sipwise kurzfristig bestehende VoIP Services ablösen und in einem nächsten Schritt auch ältere ISDN Anschlüsse auf IP Technologie migrieren.“ erklärt Gernot Fuchs, COO von Sipwise. „In Zukunft ermöglicht die flexibel erweiterbare Plattform auch die Einführung neuer Produkte und Funktionalitäten wie etwa virtuelle Nebenstellenanlagen für Firmenkunden oder Fixed Mobile Convergence für alle Anschlüsse.“ Damit ist NetCologne bestens für die aktuellen und zukünftigen Herausforderungen des Telekommunikationsmarktes gerüstet.
Über NetCologne
NetCologne zählt mit mehr als 385.000 Telekommunikationskunden und über 215.000 Kunden für TV-Kabelnetzdienste zu den erfolgreichsten Regionalcarriern Deutschlands. In den letzten Jahren setzt das Unternehmen verstärkt auf die Schaffung von breitbandigen Infrastrukturen in seinem Verbreitungsgebiet. Im Stadtgebiet Köln hat NetCologne ein hochleistungsfähiges Glasfasernetz auf Basis der Technologie „FttB“ (Fibre to the Building) verlegt. Außerhalb des Stadtgebiets bietet das Unternehmen mittels der Ausbautechnik „FttC“ (Fibre to the Curb, dem Glasfaseranschluss bis an den Kabelverzweiger) in vielen Regionen Breitbandversorgung an – entweder durch eigenen Netzausbau oder im Rahmen von Kooperationen. Zu diesen Ausbaugebieten zählen unter anderem Windeck, Burscheid, Betzdorf, Niederkassel, Pulheim, Wesseling und Eitorf. Insgesamt umfassen die Gebiete ein Potenzial von knapp 80.000 Haushalten und Gewerbebetrieben.
Über Sipwise
Sipwise GmbH ist in Europa der führende Anbieter für innovative Telekommunikationslösungen der nächsten Generation. Das Unternehmen bietet professionelle „Voice over IP“ und Unified Communication (UC) Lösungen auf Hardware- und Softwarebasis an. Neben namhaften Telekom-Anbietern wie Liberty Global UPC gewinnt Sipwise immer mehr große, mittlere und kleinere Netzbetreiber als Kunden. Zuletzt kamen vor allem auf dem deutschen und dem Schweizer Markt zahlreiche renommierte Unternehmen als Neukunden hinzu. Sipwise wurde 2008 gegründet und hat seinen Firmensitz in Brunn am Gebirge, Österreich.
Ihr Ansprechpartner:
Sipwise GmbH, Susanne Windisch
Europaring F15, 2345 Brunn am Gebirge
Email: swindisch@sipwise.com, Office: +43(0)130120-12, Internet: www.sipwise.com
Sipwise will exhibit its current NGC product portfolio at Africa´s largest Digital Business Exhibition & Conference.
Cape Town/ Brunn am Gebirge, September 03, 2014
AFRICAcom is considered being Africa´s leading digital business platform and will take place from November 11-13 in South Africa. Already for the 17th time the annual event provides a platform for telecoms, media and the ICT market in Africa. More than 375 exhibitors provide some 9,000 attendees from more than 140 countries with a glimpse of new trends in the digital industry. Check out www.comworldseries.com/africa for further information.
“With its product range, Sipwise offers the most eligible VoIP solution for Africa” feels Daniel Tiefnig, CEO of Sipwise, confident and explains “the easy installation of our products, their flexibility and the affordable price – that is great value for money!” Existing customers on the continent second that. During the exhibition, Sipwise will demonstrate the company’s latest concepts, products, and services that put this philosophy into reality.
Exhibition attendees will have the opportunity to test and try the fixed line and mobile solutions on-site. Visit us at booth A17/18 and follow us on twitter or facebook.
About Sipwise
Sipwise is the leading provider of innovative telecommunication solution of the next generation. The company offers professional Voice over IP and Unified Communication (UC) solutions on hard- and software basis. Lately Sipwise introduced successfully its “NGN in a box” solution providing cost effective and simple access to Deutsche Telekom’s new SIP-based NGN interconnect N-ICA.
Sipwise was founded in 2008 in Austria and operates globally with currently twenty employees. Development, consulting and sales are covered under one roof.
For questions and further information, please contact:
Sipwise GmbH, Susanne Windisch
Europaring F15, 2345 Brunn am Gebirge
Email: swindisch@sipwise.com, Office: +43(0)130120-12, Internet: www.sipwise.com
We are excited to announce the general availability of sip:providerCE mr3.4.2 and sip:providerPRO mr3.4.2, aka the next build of release v3.4 Version.
The Sipwise sip:provider platform is a highly versatile open source based VoIP soft-switch for ISPs and ITSPs to serve large numbers of SIP subscribers. It leverages existing building blocks like Kamailio, Sems and Asterisk to create a feature-rich and high-performance system by glueing them together in a best-practice approach and implementing missing pieces on top of it.
Sipwise engineers have been working with Asterisk and Kamailio (and its predecessors SER and OpenSER) since 2004, and have roles on the management board of Kamailio and are contributing to these projects both in terms of patches and also financially by sponsoring development tasks. The sip:provider platform is available as a Community Edition (SPCE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup.
The SPCE provides secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, file transfer, screen sharing, remote desktop control) and connect them to other SIP-, Mobile- or traditional PSTN-networks. It can therefore act as open Skype replacement system, traditional PSTN replacement, Over-The-Top (OTT) platform and also as a Session Border Controller in front of existing VoIP services in order to enable signaling encryption, IPv6 support, fraud- and Denial-of-Service prevention. Another use-case is to act as a Class4 SIP concentrator to bundle multiple SIP peerings for other VoIP services.
The build mr3.4.2 contains the set of fixes for release mr3.4. No new features added to mr3.4.2. See the full list of changes at the end of the announce.
As usual, we’re providing a VMWare Image, a Virtualbox Image and a Vagrant Box for quick evaluation testing. For those of you using Amazon Cloud we provide the EC2 AMIs in the following regions:
Check the relevant section in the Handbook for detailed instructions.
For new users, please follow the Installation Instructions in the Handbook to set up the SPCE mr3.4.2 from scratch.
For the users of the previous version of the SPCE, please follow the upgrade procedure outlined in the Handbook. If you have customized your configurations using customtt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise all your calls will most certainly fail.
Over the last months we’ve started to publish our software components at github.com/sipwise. This is still an on-going effort, which is done on a component-per-component basis. Please check back regularly for new projects to appear there, and feel free to fork them and send us pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at lists.sipwise.com/listinfo/spce-dev.
The mr3.4.2 build is stabilization/bugfixes build, so contains no new features. We are preparing some exciting new features for the upcoming mr3.5.1. Also, there will be some further enhancements to the REST-API, so any feedback on the current state is highly appreciated, as it is the first version we consider stable and feature-complete. The SOAP/XMLRPC API does not receive any new features anymore, so this is a good time to consider switching to the new API.
We want to thank our PRO customers and the SPCE community for their feedback, bug reports and feature suggestions to make this release happen. We hope you enjoy using the mr3.4.2 build and keep your input coming. A big thank you also to all the developers of Kamailio, Sems and Prosody, who make it possible for us to provide an innovative and future-proof SIP/XMPP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise development team, who worked insanely hard to create this release. You are awesome!
MT#9085 $avp(s:acc_state) not always set
MT#9003 upgrade problem still exists after changing SPCE hostname
MT#8981 Non descriptive error on wrong user name in subscribers put api request
MT#8915 Double Record_route entry in INVITE after Failure Route cause ACK getting lost
MT#8873 ngcp-panel subscriber dashboard doesn’t show number of recent calls
MT#8871 Prepare and release mr3.4.2 (release mr3.4 build 2)
MT#8859 Fix $rU, $tU handling on blind transfer
MT#8793 After upgrade to 3.4, new preference “music_on_hold” is set to 0 by default
MT#7007 Use dom/usr outbound_* preferences when force_outbound_calls_to_peer=1
MT#8741 Remove unnecessary [- IF PRO -] checks in case of using ngcp-check_active
MT#8711 review contact masking using advertised address
MT#8709 API REST: Deletion of a subscriber return an ERROR 500
MT#8707 optimize EC2 AMI for public usage
MT#8699 kamailio: debugger reset_msgid fails when the message has been processed already
MT#8653 Unable to register to peer via sems reg_agent
MT#8649 callforwards link missed in subscriber api info
MT#8633 is_primary flag in provisioning.voip_dbaliases not set when creating new subscriber
MT#8621 0008559: CallFowardings give Internal Server Error (incident 1388773ED7B6F750)
MT#8609 REST API: CFT – description and half empty objects after deleting.
MT#8563 Fix style errors and typos in documentation
MT#8529 sems/sems-pbx fails to compile against Debian/jessie
MT#8523 Present alias number in edr
MT#8449 Missing entry in kamailio.dbaliases if subscriber is created via REST API
MT#8441 CE ngcp-eaddress moves ssh_ext to $IFACE, so we cannot connect SSH to previous address anymore.
MT#8433 Invoice does not show all information
MT#8401 Encoding problem with special characters in invoice generation
MT#8387 There seems to be a rounding problem in invoice pdf totals
MT#8367 rest api: Mysql error on reseller login
MT#8359 disable INFO support in Allow header if it’s not supported
MT#8353 Increase number of digits after the decimal point for printed invoices
MT#8329 add allowed_clis as customer preference
MT#8309 Get rid of JSON::Types::bool()
MT#8225 HA not working for Cloud PBX [PRO only]
MT#8195 rate-o-mat selecting wrong billing profile if peers have same host name
MT#8067 create /etc/ngcp_nodename
MT#7867 Logo inside PDF invoice can’t be changed
MT#7769 CloudPBX: NOTIFY message has a CALLID not compliant [PRO only]
MT#6299 Challenge domain deletion with security dialog in old www_admin panel
MT#5879 Improved PDF Invoice generation
We are excited to announce the general availability of sip:providerCE mr3.5.1 and sip:providerPRO mr3.5.1, aka the new v3.5 Version.
The Sipwise sip:provider platform is a highly versatile open source based VoIP soft-switch for ISPs and ITSPs to serve large numbers of SIP subscribers. It leverages existing building blocks like Kamailio, Sems and Asterisk to create a feature-rich and high-performance system by glueing them together in a best-practice approach and implementing missing pieces on top of it.
Sipwise engineers have been working with Asterisk and Kamailio (and its predecessors SER and OpenSER) since 2004, and have roles on the management board of Kamailio and are contributing to these projects both in terms of patches and also financially by sponsoring development tasks. The sip:provider platform is available as a Community Edition (SPCE), which is fully free and open source, and as a commercial PRO appliance shipped turn-key in a high availability setup.
The SPCE provides secure and feature-rich voice and video communication to end customers (voice, video, instant messaging, presence, buddy lists, file transfer, screen sharing, remote desktop control) and connect them to other SIP-, Mobile- or traditional PSTN-networks. It can therefore act as open Skype replacement system, traditional PSTN replacement, Over-The-Top (OTT) platform and also as a Session Border Controller in front of existing VoIP services in order to enable signaling encryption, IPv6 support, fraud- and Denial-of-Service prevention. Another use-case is to act as a Class4 SIP concentrator to bundle multiple SIP peerings for other VoIP services.
The most important changes for mr3.5.1 are:
As usual, we’re providing a VMWare Image, a Virtualbox Image and a Vagrant Box for quick evaluation testing. Check the relevant section in the Handbook for detailed instructions. Also you can use our AMI (Amazon Machine Images) image or Docker container. For those of you using Amazon Cloud we provide the EC2 AMIs in the following regions: region us-east-1: ami-c647ffae , region us-west-2: ami-abe6aa9b , region us-west-1: ami-055f4a40 , region eu-west-1: ami-7e54f809 , region ap-southeast-1: ami-649fb936 , region ap-southeast-2: ami-15462b2f , region ap-northeast-1: ami-610d3b60 , region sa-east-1: ami-3da41020 .
For new users, please follow the Installation Instructions in the Handbook to set up the SPCE mr3.5.1 from scratch.
For users of the SPCE mr3.4.x, please follow the upgrade procedure outlined in the Handbook. If you have customized your configurations using customtt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise all your calls will most certainly fail.
We are publishing our software components at github.com/sipwise. This is still an on-going effort, which is done on a component-per-component basis. Please check back regularly for new projects to appear there, and feel free to fork them and send us pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at lists.sipwise.com/listinfo/spce-dev.
We want to thank our CARRIER and PRO customers and the SPCE community for their feedback, bug reports and feature suggestions to make this release happen. We hope you enjoy using the mr3.5.1 release and keep your input coming. A big thank you also to all the developers of Kamailio, Sems and Prosody, who make it possible for us to provide an innovative and future-proof SIP/XMPP engine as the core of our platform! And last but not least a HUGE thank you to the Sipwise development team, who worked insanely hard to create this release. You are awesome!
MT#9581 SLA NOTIFY is not routed properly in Carrier 3.x
MT#9579 Carrier 3.x Database.central.dbhost must be pointed to db01
MT#9573 Carrier 3.x Update mysql perms for sp1/sp2
MT#9569 Empty user in location table generate invalid INVITE for SEMS
MT#9515 Sems-pbx reply 488 Invalid Fingerprint in mr3.4.2 to call to auto-attendand
MT#9481 Add credit via REST API
MT#9479 Carrier 3.x NGCP related cron jobs are being installed and executed on all the nodes
MT#9469 Monit restarts nginx if ngcp-panel is not running
MT#9445 Voicemail notification report wrong time
MT#9441 Fixed Call Forwarding to external Voice-Mail
MT#9439 Multiple rewrites of Allow header produce malformed heaeder
MT#9431 Huntgroup policy provisioned via API seems not properly applied in DB
MT#9427 ngcpcfg apply produce an error on inactive node during upgrade (ERROR: connect_unix_sock: connect(ctl.proxy.sock): No such file)
MT#9425 monit 5.9 failed to start some components in trunk (proxy,sems,prosody,ngcp-panel) due to wrong umask on /tmp/XXX folders
MT#9423 Improve ngcp-ppa usability
MT#9419 Remove Diversion from inbound message
MT#9407 Carrier 3.x Add Elasticsearch old indexes cleanup via cron
MT#9403 Voicebox: play empty-prompt when changing to empty folder
MT#9397 Carrier 3.x check_mysql_repl cannot connect database on non-proxy servers
MT#9393 /etc/ngcp-monitoring-tools/collective-check.conf has mysql check on DB node only
MT#9385 Carrier 3.x obtain a unique dbnode
MT#9379 Modifications on db01b are not replicated on the slaves
MT#9377 Carrier 3.x Fixed MWI for carrier case
MT#9373 Kamailio cannot start, low PKG_MEMORY:<core> [io_wait.c:516]: init_io_wait(): ERROR: init_io_wait: could not alloc fd array ()
MT#9369 Export primary_number and alias_numbers in device config
MT#9365 Carrier 3.x Elasticsearch 1.3.4 should be installed by default
MT#9333 Improve debug output for tt2-daemon cannot create socket (error: can’t setup server at /usr/share/ngcp-ngcpcfg/helper//tt2-daemon line 37)
MT#9331 Improve error checking for call to Mobile Push
MT#9329 Call Dsitribution graph – timeout with less then 1 million entries
MT#9319 Fixed JS errors on ngcp-panel in IE9
MT#9313 Document early_reject sound prompts
MT#9307 Ngcp-panel: broken invoice-gen cron job
MT#9299 Single PRO node doesn’t became active on inital boot (HA ERROR: Return code 255 from /etc/init.d/redis-master)
MT#9295 Ngcp-panel: avoid external jquery.min.js resource inclusion in our source code
MT#9291 API Callforwards implement uri deflation
MT#9285 Carrier 3.x Redesign config.yml to be equel on all hosts.
MT#9281 Carrier 3.x Redesign network.yml to be equel on all hosts.
MT#9269 Fixed REST API for pbxdevicemodels to be compatible with 3.4
MT#9263 REST API: cfdestinationsets has incomplete documentation and wrong properties
MT#9253 Pushd uses wrong syslog severities
MT#9251 Installer: replace dependency on aptitude with script to install aptitude
MT#9247 Sshd template: fall back to ListenAddress on all IPs if we got localhost only
MT#9239 Rewrite rule – replacement string validation
MT#9237 Advanced CF config only shows first active mapping entry
MT#9229 Carrier 3.x Proposal to change ‘ds_select_dst’ flag from hash with callid to something else for REGISTER messages
MT#9225 Add numbers to allowed_clis if ossbss.provisioning.auto_allow_cli is set
MT#9213 Re-writing Rules after edit is moved up/down if were moved up/down just before editing
MT#9209 introduce preference allowed_clis_reject_policy
MT#9205 Carrier 3.x Sometimes registration get 200 OK but no entry in location neither in kamailio mem
MT#9189 Update kamailio to 4.1.6
MT#9171 Package captagent for Debian
MT#9169 Carrier 3.x With bonding interface SNMP log daemon.debug snmpd error on subcontainer ‘ia_addr’ insert (-1)
MT#9167 Fixed contract timestamps on create/terminate
MT#9113 Fixed some minor problems during Carrier 3.x installation
MT#9085 $avp(s:acc_state) not always set
MT#9067 Db connection pooling for kamailio-lua
MT#9065 Rtpengine package upgrade ordering
MT#9051 Lookup callee via PBX extension instead of RWR
MT#9047 Remove voip_allowed_ip_groups when terminating subscriber
MT#9041 sipwise-kamailio-config-tests starts failing after upgrade CE mr3.4->trunk since Sep 8
MT#9023 Fixed Unattended/Blind transfer for some corner cases.
MT#9003 Fixed upgrade problem after changing SPCE hostname
MT#8995 Removed hardcoded sp1/sp2 from snmpd.conf.tt2
MT#8981 Non descriptive error on wrong user name in subscribers put api request
MT#8969 cdr-exporter: fixed perlcritic errors
MT#8967 Carrier 3.x Fixed voicemail/faxserver/appserver routing
MT#8961 Fixed auth.log, contained tons of pam_env(cron:session): Unable to open env file: /etc/default/locale: No such file or directory
MT#8959 [ngcp-panel] correct peering server’s weight field validation
MT#8951 Add numbers and names to prosody sipwise-vcard module
MT#8915 Double Record_route entry in INVITE after Failure Route cause ACK getting lost
MT#8911 Carrier 3.x will have the same software installed on all the nodes.
MT#8903 Carrier 3.x Monit starts unnecessary services on inactive node (with status Execution failed)
MT#8901 redis-server has sed -e which aborts init script execution in case on non-zero ngcp-check_active exit code
MT#8895 Moving hylafax and iaxmodem under monit
MT#8889 SP2 on newly installed PRO-trunk doesn’t have role RTP (Monit doesn’t monitor rtpengine).
MT#8873 ngcp-panel subscriber dashboard doesn’t show number of recent calls
MT#8859 Fix $rU, $tU handling on blind transfer
MT#8851 xmpp pushd support
MT#8833 Prosody init script is outdated in templates (missed support of /etc/default/prosody)
MT#8831 Nginx init script from templates is outdated
MT#8829 Apache init script from templates is outdated and has a typo
MT#8827 Collectd init script from templates is outdated
MT#8823 ngcp-panel – create special offpeak date error on validate
MT#8809 ngcp-upgrade – renaming customtt.tt2 to customtt.tt2.ngcp-old – only once
MT#8797 w_record_route(): Double attempt to record-route
MT#8793 Fixed new preference music_on_hold after upgrade to 3.4 (0 by default)
MT#8777 templates – move logrotate.d / glusterfs -> system
MT#8773 ngcpcfg-api: adapt api to Carrier 3.x
MT#8767 Use peer rewrite rules for calls forced to peer by pref.
MT#8763 ngcpcfg – extend to cover multi template paths
MT#8753 kamailio-config-tests: avoid changes on already configured files
MT#8745 CE has no rc*.d symlinks
MT#8743 Improve ngcp-backup stability
MT#8741 Remove unnecessary [- IF PRO -] checks in case of using ngcp-check_active
MT#8739 glusterfs installation fails on Debian/jessie
MT#8735 Fraud lock e-mail lists terminated subscribers
MT#8729 Failing api test
MT#8727 Please add ngcp-check_active as a link to true for CE
MT#8723 CloudPBX: provide direct link for firmware download
MT#8721 Kamailio-config-tests: responder sipp pid are not correct
MT#8715 Prevent system to start more than one ngcp-www_csc daemon
MT#8711 Review contact masking using advertised address
MT#8709 API REST: Always update lock-level
MT#8707 Optimize EC2 AMI for public usage
MT#8703 Provide facilities to check NGCP code and default configuration integrity: ngcp-status
MT#8699 kamailio: debugger reset_msgid fails when the message has been processed already
MT#8673 uninitialized variables are used in ROUTE_EARLY_REJECT
MT#8663 PRO: redis.lua:764: could not connect to localhost:6379
MT#8659 Nginx fails to install on Debian/jessie
MT#8657 EC2 AMI: improve setup for lower performance systems
MT#8653 Unable to register to peer via sems reg_agent
MT#8649 callforwards link missed in subscriber api info
MT#8639 Carrier 3.x update admin_telnet config
MT#8633 is_primary flag in provisioning.voip_dbaliases not set when creating new subscriber
MT#8627 Implement checking URI field in simple callforwards form
MT#8623 Do not brake the replication if repuser password changes
MT#8621 Check callforward spec against empty desctinations
MT#8609 REST API: CFT – description and half empty objects after deleting.
MT#8591 With sst_enable=no, Session-Expires, Min-SE are not propagated but Supported: timer yes.
MT#8589 installer: check for cdrom entry in sources.list and warn user
MT#8577 system-tools: provide make target to check for syntax errors
MT#8567 Allow to supress file changed as we read it in ngcp-backup
MT#8563 Fix style errors and typos in documentation
MT#8561 Filter write methods in API for r/o users
MT#8559 CallFowardings: dapt URI field helpers to empty value (on creation).
MT#8553 Re-INVITE from upstream breaks masking contact
MT#8549 PRO: hb_watchdog, sshd connection limits and sshd_config.tt2
MT#8547 ngcp-panel/ngcp-schema: use of given and Smartmatch breaking with Perl >=5.18
MT#8545 Apache 2.4 expects sites as $site.conf instead of $site
MT#8527 collectd-mod-redis: uses wrong include path for collectd-dev in Debian/jessie
MT#8523 Use alias number for edr export.
MT#8521 Introduce: role rtp in network.yml (supporting proxy and rtpproxy on separate hosts)
MT#8517 Carrier 3.x kamailio: lookup always on DB
MT#8507 ngcp-panel: get rid of libdigest-sha3-perl usage
MT#8503 Add cluster_set section for controlling multiple lbs and rtpengines
MT#8495 banlist on lb via xmlrpc can return another level of nesting
MT#8491 Fixed ability to terminate subscriber via REST – get 500
MT#8481 Improve ngcp-upgrade behaviour re mysql db dumps
MT#8477 service ngcp-rate-o-mat stop produces: Can’t call method finish on an undefined value at /usr/sbin/rate-o-mat line 1495
MT#8475 asterisk doesn’t listen at the sip_int shared IP
MT#8463 HA node failed to became active (HA ERROR: Return code 255 from /etc/init.d/redis-master)
MT#8449 Missing entry in kamailio.dbaliases if subscriber is created via REST API
MT#8447 redis: configure persistence
MT#8441 CE ngcp-eaddress moves ssh_ext to $IFACE, so we cannot connect SSH to previous address anymore.
MT#8437 Discrepancy between CSC Web password field and Admin Web password field
MT#8433 [mkth] Invoice does not show all information
MT#8419 use /etc/nodename not hostname generally
MT#8409 Introduce redis on CE
MT#8401 Encoding problem with special characters in invoice generation
MT#8397 Remove usage of libsys-sig-perl, libregexp-parser-perl, libmoosex-fileattribute-perl
MT#8395 Remove usage of libcatalyst-plugin-email-perl
MT#8387 There seems to be a rounding problem in invoice pdf totals
MT#8371 Customer search is terribly slow in case of big customer’s amount
MT#8367 REST API: Mysql error on reseller login
MT#8359 Add SIP INFO support
MT#8353 Increase number of digits after the decimal point for printed invoices
MT#8329 Add allowed_clis as customer preference
MT#8325 Fixed ngcp-terminate_subscriber
MT#8311 Carrier 3.x add support to ngcp-collective-check
MT#8309 Get rid of JSON::Types::bool()
MT#8299 CloudPBX: implement visual key/line selection in web interface
MT#8297 Start SSH server in reboot after installation grml process
MT#8287 Fixed minor issues with PBX extension management
MT#8279 User input validation for CF target numbers in new admin panel seems to be broken.
MT#8271 Carrier 3.x introduce mgm_int interface type
MT#8261 REST API: api/cftimesets unable to create time set with minutes interval starting with ’0′
MT#8259 mediaproxy session is not destroyed in ROUTE_STOP_RTPPROXY_BRANCH
MT#8247 Search sipstats by UUID fails
MT#8243 dlg counters in central redis
MT#8225 HA not working for Cloud PBX
MT#8215 Research central log storage options for carrier 3.x open
MT#8203 Registered Devices are not deleted on subscriber termination
MT#8195 Rate-o-mat selecting wrong billing profile if peers have same host name
MT#8193 Use proper firmware for SPA512G and SPA514G phones
MT#8181 Sems 1.6: db_reg_agent functions are not exported to XMLRPC API
MT#8171 asterisk/ngcpcfg.services doesn’t check active node
MT#8167 Move sp*-var-lib-glusterfs-export.pid to /var/run
MT#8157 REST API: Callforward destinations are removed when clearing time sets
MT#8155 ngcp-sync-constants needs to connect to the right server and sync the proper grants to it
MT#8153 ngcpcfg push produces an error even with packages that are not installed.
MT#8127 Carrier 3.x Multiple instance of mysql service
MT#8125 Carrier 3.x Run services depending the role the node has open
MT#8121 Incoming calls interrupted afer 90-180 seconds: fix adding P-Out-Socket to reply (remove ambigous checks)
MT#8117 Carrier 3.x Introduce: role db
MT#8093 Fix instructions for downloading ngcp-installer in 3.x handbooks
MT#8071 Unable to re-assign a number of a terminated subscriber
MT#8067 Create /etc/ngcp_nodename
MT#8065 Problem with billing profile changing
MT#8063 API REST: api/soundfiles unable to upload MOH sound file
MT#8035 Write start_ivr and end_ivr events
MT#8031 ngcpcfg-api: migrate from apache to nginx
MT#7995 lcr_rate: ignore peers without valid fees
MT#7985 Allow ntp listen all interafces but restrict to ntp.servers from config.yml
MT#7943 API: prevent accessing provisioning.voip_subscriber fields on termination if it doesn’t exist anymore
MT#7931 Missed grouping on captured dialogs section of subscriber page for voice calls
MT#7927 Cannot SSH system using cdrexport account (su: Cannot determine your user name.)
MT#7867 Logo inside PDF invoice can’t be changed
MT#7861 Invoice template creation: Use pixels as main unit to be in accordance with svg-edit.
MT#7843 VAT Rate and VAT included missing from Billing Fees
MT#7805 Commit noatime by default to improve system performance
MT#7803 Show invoice links and sections in panel
MT#7791 ngcp-panel: no callthrough CLIs configuration page
MT#7705 Increase the oss.log. No significant info provided right now
MT#7673 Possibility to set the Voicemail number
MT#7607 glusterfs – Fixed logs rotation issue
MT#7573 Upgrade script will cleanup old core files
MT#7533 kamailio-config-tests: stop rate-o-mat service during kamailio tests
MT#7335 Debian/jessie issues in ngcp and related packages
MT#7331 PRO: Update Hylafax+ package to latest upstream release
MT#7327 add jessie support in daily builds/Jenkins jobs
MT#7319 RTP bandwidth limits and optimizations for calls within same network
MT#7275 templates/lsb_scripts are out of sync with debian/${init} scripts
MT#7177 Rewrite numbers for subscriber roles in panel based on its rewrite rules
MT#7007 Use dom/usr outbound_* preferences when force_outbound_calls_to_peer=1
MT#6969 ngcp-panel: trigger SPA device resync via SIP NOTIFY
MT#6485 Improve docker images for sip:provider CE
MT#6299 Challenge domain deletion with security dialog
MT#5879 Improving PDF Invoice generation
MT#5599 Rework ngcp-panel auth/authz to support password-based API access