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Sip:provider mr3.4.1 Is Now Available on Sipwise Website!

Sipwise released the latest Sip:provider mr3.4.1

We are excited to announce the general availability of sip:provider mr3.4.1, CE and PRO, aka the new v3.4 Version.

What’s the Sip:provider platform?

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 connects 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 signalling 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.

What’s new in the Sip:provider mr3.4.1?

The most important changes for the Sip:provider mr3.4.1 are:

  • General System Fixes and Enhancements
  • NGCP-Panel
    • Enhanced Subscriber Profile Handling
    • Improved PDF invoice generation
    • Event exporter allows you to charge customers based on the features they are using instead of the calls
    • A number of the ngcp-panel, as well as www_csc processed, can be configured in config.yml (search for fastcgi_workers)
    • Feature-Completion of the REST-API – many new resources available
    • Updated translations for German, Italian, Russian
  • SIP/RTP Core Enhancements
    • Kamailio is updated to 4.1.4
    • Sems receives an update to 1.6 (CE only)
    • Add Kamailio reload after updating lua-ngcp-Kamailio package
    • Implement History-Info according to RFC44224 and Deutsche Telekom requirements
    • New preference bypass_rtpproxy to allow media to bypass rtpengine for users behind same NAT
    • More fine-grained call admission control – the concurrent_max and concurrent_max_out counters do not include calls to voicemail and application server
    • Introduced the new set of preferences concurrent_max_total, concurrent_max_out_total that set total limit including calls to voicemail and application server
    • Find voicemail user by alias when querying the mailbox externally
    • New preference voicemail_echo allows changing voicemail mailbox number in web interface
    • New security mechanism to prevent in-dialog requests on caller’s leg from bypassing proxy
    • Add preference allowed_clis on the Customer level
    • New preference force_strict_number_match – disallow dialling arbitrary extensions behind main subscriber number
    • Rtpengine – configurable per-call TOS value
    • Rtpengine – better ICE priority calculation for non-RFC clients
    • Remove REFER if allow_refer_method is disabled
    • Remove INFO if allow_info_method is disabled

How do I test-drive the new version?

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.

How do I install the new version or upgrade from an older one?

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 customised your configurations using custom tt.tt2 files, you must migrate your changes to the new configuration files after the upgrade, otherwise, all your calls will most certainly fail.

How can I contribute to the project?

Over the last months, we’ve started to publish our software components at 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 to pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at


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!

Full Changelog of Bugfixes and Enhancements

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/ doesn’t check active node
MT#8147 Prepare and release mr3.3.2 (release 3.3 build 2)
MT#8121 Incoming calls interrupted after 90 upto 180 seconds (fix adding P-Out-Socket to reply, remove ambiguous 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 any 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 username.)
MT#7901 Upgrade 3.1=>mr3.2.2: strange behaviour 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 call forwards
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 redirects 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 Attendant Slot menu is available for Standard 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 calling card 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 Attendant destination  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 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