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sip:provider mr4.0.2 Released

We are excited to announce the general availability of sip:providerCE mr4.0.2 and sip:providerPRO mr4.0.2.

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 mr4.0.2?

This is the last build of mr4.0 release series. This includes only fixes from mr4.0 series. mr4.0 is out of support, please upgrade to the latests mr4.1 version available.

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-951153ff
  • AMI ID for region us-west-2: ami-66130107
  • AMI ID for region us-west-1: ami-97f39af7
  • AMI ID for region eu-central-1: ami-c3021faf
  • AMI ID for region eu-west-1: ami-7056f103
  • AMI ID for region ap-southeast-1: ami-e9b2728a
  • AMI ID for region ap-southeast-2: ami-d2376fb1
  • AMI ID for region ap-northeast-1: ami-dd2f00b3
  • AMI ID for region sa-east-1: ami-d771f4bb

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 mr4.0.2 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.

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 mr4.0.2 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!

Full Changelog of Bugfixes and Enhancements

MT#9759 Add pre-upgrade checks to ngcp-status
MT#8457 Easy way to duplicate billing profiles
MT#2161 Add user=phone param for calls from/to phone numbers
MT#16587 Brocken UTF encoding in field
MT#16369 Save cores by default
MT#16367 fix voisniff scripts (rotate-partitions.pl)
MT#16365 Reverse Media Negotiation not working when first INVITE is without RTP - rtpengine not in mediapath
MT#16341 Brocken UTF encoding sent to Google Push server (GCM)
MT#16329 user=phone needed in the From Header for Enviatel peer customer Suc Dacour New
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 clounters that are unused for e.g. 24 hours
MT#16113 bad translation
MT#16083 Problem setting outbound_pai_user when profile_preferences are involved
MT#16057 Fix DBIx::Class syntax for searching numbers in /api/interceptions
MT#15919 CLIR not kept is destination number is a queue
MT#15903 ngcp-installed failed to install spce on debian plain if “ca-certificates” is not installed
MT#15805 BLF funtion should not be available on Panasonic KX-UT113/123
MT#15803 check pull before ngcpcfg apply
MT#15757 Not possible to disable Voicemail notification via Email
MT#15745 Call Lists - Change the display in the following cases: Call Transfer
MT#15701 REST API create/delete domain calls xmpp/SIP before commiting changes to DB
MT#15685 Panel get 404 Not Found when editing Rewriting Rules
MT#15657 libinewrate-binaries job fails because of lintian error
MT#15651 debug package for libinewrate
MT#15593 Rtpengine DKMS kernel module failed to be installed (added only to dkms)
MT#15583 Subscriberadmin cannot access Customer Details page
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 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#15413 Push not received if multiple tokens in DB from the same account and from the same device
MT#15397 Mediator on carrier deadlocks each other  on stats update on central server
MT#15381 mediator mr3.6+ doesn't recognize accoutning from mr3.5
MT#15367 panel shows wrong zones
MT#15361 Update MoH - Old file is still played
MT#15359 CLI Presentation issue'swith CFU
MT#15341 When calling to AA the subscriber see from 00311 displaying in second leg call.
MT#15311 sems-pbx issue with PRACK when UAC/UAS are both sems
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#15247 Always force https in autoprov config
MT#15237 interfaces.yml should contain interfaces from all hosts in carrier
MT#15225 Android Group Chat Query
MT#15203 PBX group subscriber cannot be edit
MT#15123 ACK goes in loop in a IPV6 call
MT#15119 Add ability to trace XMPP connection details for perticular user
MT#15117 pickup on yealink t48g does not work on BLF button
MT#15107 queue_greeting and queue_waiting_music audio file is not playing after uploading a new audio file
MT#15079 Line/Key Range incorrect for Cisco SPA501G
MT#15075 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#15025 CLIR set, but call with *31* does not diable CLIR
MT#14989 registrar xavp_rcd is not set after registered() check
MT#14979 REST direction filter not working
MT#14977 BLF turns green during running call when duration >90 seconds
MT#14967 kamailio-config-tests: return error but TAP has no error
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#14921 Add call-id to pushd log to simplify debugging.
MT#14901 https://X.X.X.X:1443/login/reseller  outdated link
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 faoilover case
MT#14817 SRTP does not work if callee/caller are PBX users
MT#14803 The REST API returns an internal error. Even if the requested entry is not existing, a HTTP error
MT#14767 Backport  Innovaphone provisioning WITHOUT LDAP Phonebook part to mr3.8.3+
MT#14747 Fix Yealink W52P base firmware fetching
MT#14741 Fix /api/trustedsources/ when called with reseller creds
MT#14739 REST API returns
MT#14729 Enhance labels and front images for SIP-T28P + EXP39 & SIP-T28P + 2xEXP39 phone models
MT#14715 Return UUID in /api/subscribers/
MT#14703 libinewrate and charging long duration calls
MT#14681 Prepare and release mr3.8.3
MT#14673 Pushd call failed randomly (LB loops B invite on itself due to wrong P-D-URI)
MT#14667 Kamailio doesn't update presentety for expired locations
MT#14653 I am having some REST API problems!
MT#14641 Cisco Telephone restarting all the time - SPA514G
MT#14635 improve mobile push options for carrier
MT#14625 PBX Call Queues impovements
MT#14597 Bug NGCP Dashboard - Can't change E164 number of the client
MT#14559 Deleted User still register at peer through remote authentication
MT#14549 Voisniff makes NGCP unreliable!
MT#14485 Double reported entries on the CDR
MT#14479 ngcp-status: provide support for checking ntp setup
MT#14477 REST API lacking method to send and retrieve fax
MT#14461 REST API lacking method to send and retrieve fax
MT#14445 billing fees upload is broken
MT#14405 Failed SIP registration on app
MT#14395 push notification sent before cheking prepaid credit
MT#14391 panel deletes certain alias numbers when editing subscriber
MT#14383 Caller number is shown in Web Interface for customer in spite of suppressed number (anonymous Calls)
MT#14365 Sound cache reload fails if PBX is not enabled
MT#14317 REST API test failed randomly due to new tests in api-balanceintervals_t
MT#14311 ngcp-sync-db: is missing the _not_replicated.up info at ngcp.db_chema
MT#14301 Billing Packages Test
MT#14281 prosody pushd module not using urlencode?
MT#14267 REST API Documentation change
MT#14243 Implement missing gaps for libswrate for balance underrun
MT#14229 fix shellscripting issues caught by *-tap-test Jenkins jobs
MT#14227 Deutsche Telekom - Falcon project (migreate to libinewrate)
MT#14159 Voucher field descriptiion is wrong
MT#14141 initial loop to PBX is using default lbrtp_set
MT#14107 how to set default language for CSC
MT#14079 Package coturn server and templates
MT#13919 CloudPBX: Integrate Innovaphone phones to PBX
MT#13917 Call Push notification failing
MT#13909 cleanup/remove section www_csc from config.yml
MT#13903 dynamic balance interval duration and start (including topup)
MT#13833 including changes for push in 3.8LTS
MT#13825 Carrier 3.x cannot start astetrisk on package upgrade if dnsmask is disabled (Debian bug #703805/#773170)
MT#13747 RegTest - Call from PSTN with CFU to PSTN not working
MT#13743 Sems (libswrate) is writing prepaid_cost table on db01 but rate-o-mat is checking that table on localhost (prx)
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#13459 Add romanian voice prompts to ngcp-prompts
MT#13367 Softclient Desktop must be in the Subscriber Profile's properties
MT#13363 Auto attendant destination not working if destination has PBX Call Queue enabled
MT#13359 No incoming call from call queue after attended call transfer
MT#13251 Customer Music On Hold file is not being played
MT#13229 New Captagent package
MT#13155 Audible ringing for calling user continues for 3 minutes even when huntgroup member ringing stops after 10 seconds
MT#13059 Auto attendant destination not working if destination has PBX Call Queue enabled
MT#13011 Fix accounting on calls to AA target
MT#13007 PBX sounds reload in System sound set is not handled correctly
MT#12985 Subscriber cannot edit
MT#12937 Order of huntgroup member in API call
MT#12879 rtpengine <-> rtpengine-redis
MT#12873 Bugs found during PBX Call Queue tests
MT#12825 Exporting button for billing fees
MT#12347 Tests rewritten without copy-paste of handreds lines of code
MT#12307 e164_to_ruri option doesn't work for PBX users
MT#11921 clean tap-test errors/warnings
MT#11831 Please deploy default audio files for langugages other than English

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