We are excited to announce the general availability of sip:providerCE mr3.8.4 and sip:providerPRO mr3.8.4.

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 mr3.8.4?

The most important changes for mr3.8.4 compared to mr3.7.x are:

  • Direct upgrade from 2.8 LTS to mr3.8.4 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-ecda9e86
  • AMI ID for region us-west-2: ami-01c6d660
  • AMI ID for region us-west-1: ami-52294032
  • AMI ID for region eu-central-1: ami-20cbd94c
  • AMI ID for region eu-west-1: ami-c2cb6eb1
  • AMI ID for region ap-southeast-1: ami-87c706e4
  • AMI ID for region ap-southeast-2: ami-f5613896
  • AMI ID for region ap-northeast-1: ami-f07b599e
  • AMI ID for region sa-east-1: ami-4415af28

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 mr3.8.4 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 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.

Acknowledgements

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 mr3.8.4 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 mr3.8.3

MT#2161 Add user=phone parameter for calls from/to phone numbers
MT#16329 user=phone needed in the From Header
MT#16369 Save cores by default
MT#16343 Direct PRO upgrade 2.8->3.8 has an unnecessary CE statement
MT#16341 Broken UTF encoding sent to Google Push server (GCM)
MT#16325 add ngcpcfg.services file to /etc/kamailio/lb/db/
MT#16323 REST API call to calllists lacking rating_status, which is available in SOAP
MT#16307 add option to disable early media in mobile_push.dsm
MT#16279 Prepare and release mr3.8.4/mr4.0.2
MT#16213 Clear redis counters that are unused for e.g. 24 hours
MT#16171 A lot of slow queries on db01 node
MT#16113 bad translation
MT#16083 Problem setting outbound_pai_user when profile_preferences are involved
MT#16075 Moving apns PUSH parameter to config.yml
MT#16057 Fix DBIx::Class syntax for searching numbers in /api/interceptions
MT#15819 Push to apple for group chat creation does not contain “Alert”
MT#15803 check pull before ngcpcfg apply
MT#15757 Not possible to disable Voicemail notification via Email
MT#15703 LAB proxy is crashing periodically
MT#15701 REST API create/delete domain calls xmpp/SIP before committing changes to DB
MT#15685 Panel get 404 Not Found when editing Rewriting Rules
MT#15593 Rtpengine DKMS kernel module failed to be installed (added only to dkms)
MT#15555 ability to set ACC flags after transaction has been created
MT#15515 no media proxy with reverse media negotiation
MT#15509 ngcp-sems and sems-pbx should clear audio cache on start
MT#15501 pushd is running on every nodes
MT#15489 0015485: REST API for sending faxes (POST /api/faxes/) has problems with German Umlauts
MT#15459 SIP password with < symbol not displayed correctly
MT#15455 IPv6 call fails because of a loop in ACK
MT#15441 API api/subscribers modify also preferences
MT#15427 captagent templates
MT#15425 audio_cache reload is not triggered after deleting file
MT#15423 captagent not in the list of services
MT#15407 backport fixes for SIP push support
MT#15397 Mediator on carrier deadlocks each other on stats update on central server
MT#15389 Errors during the switchover
MT#15381 mediator mr3.6+ doesn’t recognize accounting from mr3.5
MT#15359 CLI Presentation issues with CFU
MT#15303 Implementing DoS/DDoS attack against XMPP
MT#15289 ngcp does not send gcm registration id
MT#15261 ngcpcfg: ability to redefine general options for node-type and specific node
MT#15237 interfaces.yml should contain interfaces from all hosts in carrier
MT#15123 ACK goes in loop in a IPV6 call
MT#15025 CLIR set, but call with *31* does not disable CLIR
MT#14989 registrar xavp_rcd is not set after registered() check
MT#14979 REST direction filter not working
MT#14939 Call to fax server on Carrier fails sometimes with 403
MT#14923 Webfax on Carrier does not work out of the box
MT#14875 Kamailio is selecting wrong value for transport_protocol for peer used multiple time
MT#14843 Invite message sent to the wrong IP in failover case
MT#14803 The REST API returns an internal error. Even if the requested entry is not existing, a HTTP error “404 not found” would be expec
MT#14741 Fix /api/trustedsources/ when called with reseller creds
MT#14739 REST API returns “201 Created” but no “Location” for “POST /subscriberregistrations/”
MT#14715 Return UUID in /api/subscribers/
MT#14681 Prepare and release mr3.8.3
MT#14653 I am having some REST API problems!
MT#14549 Voisniff makes NGCP unreliable!
MT#14477 0014461: REST API lacking method to send and retrieve fax
MT#14391 panel deletes certain alias numbers when editing subscriber
MT#14311 ngcp-sync-db: is missing the _not_replicated.up info at ngcp.db_schema
MT#14107 how to set default language for CSC
MT#13917 pua_reginfo fix from 4.3 branch, fix SUBSCRIBE from mobile_push.dsm
MT#13825 Carrier 3.x cannot start asterisk on package upgrade if dnsmasq is disabled (Debian bug #703805/#773170)
MT#13747 RegTest – Call from PSTN with CFU to PSTN not working
MT#13717 ngcp-panel nightly tests failed in trunk (selenium-ruby contact search problems + 13 errors on REST API)
MT#13459 Add Romanian voice prompts to ngcp-prompts
MT#13367 Softclient Desktop must be in the Subscriber Profile’s properties
MT#13229 New Captagent package
MT#13183 Direct carrier upgrade mr3.5.1->mr3.8.x
MT#13155 Audible ringing for calling user continues for 3 minutes even when huntgroup member ringing stops after 10 seconds
MT#12985 Subscriber cannot edit “Block List for inbound calls”
MT#12977 Backport sngrep to ngcp platform
MT#12707 Feature Malicious Call Identification ( MCID )
MT#12347 Tests rewritten without copy-paste of hundreds lines of code
MT#12307 [CloudPBX] e164_to_ruri option doesn’t work for PBX users
MT#11831 [CloudPBX] Please deploy default audio files for languages other than English
MT#15919 [CloudPBX] CLIR not kept is destination number is a queue
MT#15805 [CloudPBX] BLF function should not be available on Panasonic KX-UT113/123
MT#15745 [CloudPBX] Call Lists – Change the display in the following cases: Call Transfer
MT#15361 [CloudPBX] Update MoH – Old file is still played
MT#15341 [CloudPBX] When calling to AA the subscriber see from 00311 displaying in second leg call.
MT#15311 [CloudPBX] sems-pbx issue with PRACK when UAC/UAS are both sems
MT#15247 [CloudPBX] Always force https in autoprov config
MT#15203 [CloudPBX] PBX group subscriber cannot be edit
MT#15117 [CloudPBX] pickup on yealink t48g does not work on BLF button
MT#15107 [CloudPBX] queue_greeting and queue_waiting_music audio file is not playing after uploading a new audio file
MT#15079 [CloudPBX] Line/Key Range incorrect for Cisco SPA501G
MT#15075 [CloudPBX] Call from SRTP user to RTP user has no audio at all
MT#15057 CloudPBX: strip down autoprov HTTP headers and add Last-Modified
MT#14977 [CloudPBX] BLF turns green during running call when duration >90 seconds
MT#14817 [CloudPBX] SRTP does not work if callee/caller are PBX users
MT#14767 [CloudPBX] Backport Innovaphone provisioning WITHOUT LDAP Phonebook part to mr3.8.3+
MT#14747 [CloudPBX] Fix Yealink W52P base firmware fetching
MT#14729 [CloudPBX] Enhance labels and front images for SIP-T28P + EXP39 & SIP-T28P + 2xEXP39 phone models
MT#14641 [CloudPBX] Cisco Telephone restarting all the time – SPA514G
MT#14625 [CloudPBX] PBX Call Queues improvements
MT#14485 [CloudPBX] Double reported entries on the CDR
MT#13969 [CloudPBX] Yealink provisioning does not work as expected
MT#13931 [CloudPBX] Import phone device configuration from web interface
MT#13919 CloudPBX: Integrate Innovaphone phones to PBX
MT#13873 CloudPBX: Fix Innovaphone BLF pickup
MT#13359 [CloudPBX] No incoming call from call queue after attended call transfer
MT#13251 [CloudPBX] Customer Music On Hold file is not being played
MT#13059 [CloudPBX] Auto attendant destination not working if destination has PBX Call Queue enabled
MT#13011 [CloudPBX] Fix accounting on calls to AA target
MT#13007 [CloudPBX] PBX sounds reload in System sound set is not handled correctly
MT#12937 [CloudPBX] Order of huntgroup member in API call
MT#12903 CloudPBX: Only play AA menu prompt if a destination is set for index
MT#12761 [CloudPBX] add COLP for inter-PBX calls
MT#12245 CloudPBX: Set picked up party in response to BLF pickup request
MT#11631 [CloudPBX] MOH doesn’t work on Park the call feature on A side
MT#11147 [CloudPBX] Establish predefined line sets as special type of the device models – extension