back to all Sipwise CE - News

sip:provider mr3.8.3 Released

We are excited to announce the general availability of sip:providerCE mr3.8.3 and sip:providerPRO mr3.8.3.

What’s the sip:provider platform?

sip:provider PRO Architecture Overview

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.

What’s new in mr3.8.3?

The most important changes for mr3.8.3 compared to mr3.7.x are:

  • Direct upgrade from 2.8 LTS to mr3.8.3 LTS is now possible
  • General component stability improvements and REST API enhancements
  • The RTP bridging and transport protocol (DTLS-SRTP or SDES) for rtpengine can now be configured by user preference.
  • Update Redis to latest stable version 2.8.17
  • Update Prosody to latest stable version 0.9.8
  • Move Sipwise Debian repository from 1024bit GPG key to 4096bit key
  • Improve services security (listen appropriate interfaces only)
  • CloudPBX improvements and fixes for PRO/CARRIER customers with PBX module

Is mr3.8 LTS (long time supported) release?

Yes, release mr3.8 is the current LTS release. Release mr3.8 LTS is going to be supported for 3 years.

Is it possible to upgrade directly from 2.8 LTS to mr3.8 LTS release?

Yes, the procedure also takes care of updating Debian from 6.0 to 7.0. Check the relevant section in the Handbook for detailed instructions.

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. For those of you using Amazon Cloud we provide the EC2 AMIs in the following regions:

  • AMI ID for region us-east-1: ami-dde58ab8
  • AMI ID for region us-west-2: ami-cf667bff
  • AMI ID for region us-west-1: ami-253cc661
  • AMI ID for region eu-central-1: ami-c05656dd
  • AMI ID for region eu-west-1: ami-396a4a4e
  • AMI ID for region ap-southeast-1: ami-5c303a0e
  • AMI ID for region ap-southeast-2: ami-931956a9
  • AMI ID for region ap-northeast-1: ami-1aad261a
  • AMI ID for region sa-east-1: ami-2934be34

Check the relevant section in the Handbook for detailed instructions.

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 mr3.8.3 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.

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 pull requests. For development related questions, please subscribe to our SPCE-Dev Mailing-List at


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 mr3.8.3 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!

Full Changelog of Bugfixes comparing to mr3.8.2

MT#15025 [CloudPBX] CLIR set, but call with *31* does not disable CLIR. Proxy fixed, use passed $avp(s:caller_clir)
MT#14977 [CloudPBX] Fixed BLF which turns green during running call when duration >90 seconds
MT#14939 [Carrier 3.x] Call to fax server fails sometimes with 403
MT#14843 [PRO] Invite message sent to the wrong IP in fail-over case
MT#14817 [CloudPBX] Fixed SRTP if callee/caller are PBX users
MT#14767 [CloudPBX] Add basic Innovaphone provisioning
MT#14747 [CloudPBX] Fix Yealink W52P base firmware fetching
MT#14741 [REST API] Fix /api/trustedsources/ when called with reseller credentials
MT#14729 [CloudPBX] Enhance labels and front images for SIP-T28P + EXP39 & SIP-T28P + 2xEXP39 phone models
MT#14715 [REST API] Return UUID in /api/subscribers/ along with other subscriber info
MT#14673 [Carrier 3.x] Pushd calls failed randomly (LB loops B invite on itself due to wrong P-D-URI)
MT#14653 [REST API] Various fixes for REST API, like add required field billing_profile_id to billing zone, etc.
MT#14625 [CloudPBX] Various PBX Call Queues improvements
MT#14597 Fixed ngcp-panel: Failed to update subscriber (DBIx::Class::Schema::txn_do(): Can’t call method “all” on an undefined value at /usr/share/perl5/NGCP/Panel/Controller/ line 2449
MT#14559 Deleted User was register at peer through remote authentication, fixed.
MT#14549 [PRO] Voisniff performance fixes: locking fix, additional garbage collections, don’t wait for valid SIP data indefinitely
MT#14485 Fixed double reported entries on the CDR in exported files
MT#14395 [Carrier 3.x] Call push notification sent after checking prepaid credit and funds are enough for the call
MT#14391 Fixed panel, prevent deletes certain alias numbers by mistake when editing subscriber
MT#14383 Caller number is shown in Web Interface for customer in spite of suppressed number (anonymous Calls)
MT#14365 Sems sound cache reload fails if PBX is not enabled
MT#14281 Improve Prosody pushd module to use urlencode
MT#14267 REST API Documentation change
MT#14159 Fixed voucher field description
MT#14141 Fixed initial loop to PBX. It used default lbrtp_set
MT#14125 Fixed various errors reported for direct upgrade from 2.8 LTS to mr3.8 LTS
MT#14107 Restored possibility to set default language for new CSC
MT#14079 [CloudPBX] Add possibility to install coturn server and templates
MT#14067 /usr/share/perl5/NGCP/CDR/ fails if spaces in dir name
MT#14055 [CloudPBX] Fix cisco CA for autoprov client auth
MT#13969 [CloudPBX] Fixed Yealink provisioning whch did not work as expected
MT#13919 [CloudPBX] Integrate Innovaphone phones to PBX
MT#13917 [iOS/Android] Various fixes for Call Push notification for Sipwise iOS/Android applications
MT#13875 Add libanyevent-perl to approx cache via ngcp-approx-cache-helper
MT#13873 [CloudPBX] Fix Innovaphone BLF pickup
MT#13847 [REST API] Callists preserve number after rewrite rules
MT#13827 [Carrier 3.x] upgrade should have faildack to /usr/share/ngcp-system-tools-carrier/ instead of PRO
MT#13825 [Carrier 3.x] cannot start astetrisk on package upgrade if dnsmask is disabled (Debian bug #703805/#773170)
MT#13823 check/warn/rename custom.tt2 for sems during upgrade 2.8->mr3.8
MT#13821 [Carrier 3.x] Fix asterisk to prevent starting on all Carrier nodes
MT#13815 Split IMG2020 MGM and SIP interface
MT#13813 Optimize ngcp-fraud-auto-lock performance
MT#13789 [Carrier 3.x] enabling pushd skip link creation in /etc/nginx/sites-enabled
MT#13771 [Carrier 3.x] sems-pbx was not considering node type and trying to start on all nodes
MT#13769 Pass rtp_interface of original caller in case of CF
MT#13743 Sems (libswrate) is writing prepaid_cost table on db01 but rate-o-mat is checking that table on localhost (prx)
MT#13737 RTPengine add SDP attribute ‘rtcp’ even on T.38 Re-INVITE
MT#13711 Fixed occasional “Hash Sum mismatch” errors during apt-get runs on Debian/squeeze with haproxy
MT#13701 Fixed 500 Internal server error: Can’t call method “username” on an undefined value at /usr/share/perl5/NGCP/Panel/Controller/ line 3040.
MT#13661 app_voicemail.c doesn’t retain the umask value that is read upon
MT#13659 [CloudPBX] Number Range for Pilot subscriber cannot be used during extension creation
MT#13657 Fixed administrative flag, it was impossible to change initial settings.
MT#13617 [CloudPBX] Yealink config template doesn’t enable not default lines for the Yealink T41P & T42G devices
MT#13611 subscribers own number in CSC is now shown with 00 and sip-username, and not the number. F.e. the sip-username is 10172
MT#13599 If I set CF to VM via VSC I get wrong audio prompt
MT#13597 playbalance played in loop and call cannot be cancelled
MT#13583 handbook 3.8.2: Images are shift out of the document
MT#13573 CallList incorrect for records of terminated subscriber
MT#13543 [CloudPBX] Shared Line not working, fixed.
MT#13529 Problems with suppressed numbers
MT#13487 Fixed glusterfs migration error when upgrading PRO 2.8->mr3.8.2
MT#13471 Call Lists – Caller displayed as anonymous
MT#13459 [PRO] Add Romanian voice prompts to ngcp-prompts
MT#13447 [CloudPBX] FritzBox Voicemail call detection has been improved
MT#13435 Rewrite Rules issues
MT#13393 Fixed outbound_from_user issues with Remote Authentication
MT#13389 [CloudPBX] Add initial description of Sipwise log files in documentation
MT#13369 Fixed Inconsistent background colour of device Front Images
MT#13363 Auto attendant destination not working if destination has PBX Call Queue enabled
MT#13325 Enable logging of UA and UAIP in the kamailio.log always
MT#13297 kamailio-config-tests sometimes failed because doesn’t detect some PID modifications
MT#13269 [CloudPBX] Panel and APIs return 500 – internal server error after a POST to the api/pbxdevices
MT#13251 Fixed Customer Music On Hold file which was not played
MT#13249 Fixed acc time_hires precision issue
MT#13243 [ComX] filter subscribers by webuser/password
MT#13241 Fixed warning “Exiting subroutine via next at /usr/sbin/cdr-exporter line 65”
MT#13237 Use FILES_OWNER and FILES_GROUP when creating dirs in cdrexport
MT#13231 [PRO] Fixed collectd regusers rrd is 0
MT#13213 Voicemail does not work if advertised_ip is set
MT#13007 PBX sounds reload in System sound set is not handled correctly
MT#12975 ngcp-ppa tool supports multiple PPA repositories
MT#12939 [CloudPBX] Manage PBX groups and members order through the web panel
MT#12937 [REST API] Order of huntgroup member in API call
MT#12911 Avoid early announcements when doing Call Forward hunting
MT#12905 Voicemail failed if voicemail_echo_number & VM enabled via VSC
MT#12895 [Carrier 3.x] upgrade produces wrong /etc/apt/sources.list.d/*.list
MT#12873 [CloudPBX] Various bugs fixes during PBX Call Queue testing
MT#12761 [CloudPBX] add COLP for inter-PBX calls
MT#12347 Improved REST API tests
MT#12303 Dialogic IMG 2020 configuration as SIP/SS7 and Optical
MT#12245 [CloudPBX] Set picked up party in response to BLF pickup request
MT#12025 lua-ngcp-kamailio: improve code quality
MT#11921 Clean up various shell scripts errors/warnings
MT#11831 [PRO] Deploying default audio files for languages other than English
MT#11667 Add additional information on call list log
MT#11509 fraud-lock query creates slow-query when cdr entry are high
MT#11239 dsm/mod_utils: rename the prompts for single digits after the tens
MT#11219 Load default sound files from file system and provide failback from customer to system
MT#11147 Establish predefined line sets as special type of the device models – extension
MT#11027 Continue cleanup SSL certificate handling
MT#10879 Primary number can not be set to null if it has been already set
MT#10403 [PRO] Italian ngcp-prompts translation
MT#10401 [PRO] Russian ngcp-prompts translation
MT#10399 [PRO] German ngcp-prompts translation
MT#10337 Add update/override to cloudpbx-devices tool
MT#10125 Improved configuration for push notification for calls to offline
MT#10057 Update config.yml description in handbook
MT#6557 Improve the default sipwise user config accordingly to “Sipwise Support Access” policy
MT#6483 NGCP-API: Fetching billing profiles with lots of fees causes gateway timeout