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Sipwise is looking for Business Partners in Africa

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Brunn am Gebirge, October 29, 2014

image_Africa Banner


 
Sipwise addresses customers as well as potential partners at Africa´s largest Digital Business Exhibition & Conference in Cape Town this November.

Based in and operating from Austria/Europe, Sipwise is pushing to expand its successful business model further on a global basis. “With its product range, Sipwise offers the most eligible VoIP solution for Africa” affirms Daniel Tiefnig, CEO of Sipwise and explains why: “Easy installation, flexibility, and an affordable price of our product range turned Sipwise into a global player. The Open Source approach minimizes development costs and utilizes the innovative power of the community.”

Sipwise is looking for Sales & Distribution Partners at the AfricaCom 2014

During the exhibition, Sipwise will offer compelling insights into the company’s latest concepts, products, and services to potential customers as well as interested sales and distribution partners. “We are striving for strategic partnerships in order to professionally develop this future market,” says CMO Atilla Ceylan, who is also present at the event in November.

Visitors will also given the opportunity to experience the future trends by testing the following Sipwise products on-site:

> Unified Communication Server

> Cloud PBX

> Smartphone and Desktop App

To take an active part in the future development of Sipwise, please contact sales@sipwise.com!


 

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. Development, consulting & sales are covered under one roof.

For questions and further information, please contact:

Sipwise GmbH | Europaring F15 | 2345 Brunn am Gebirge | Email: sales@sipwise.com

 


Securing your NGCP against SIP attacks

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Sipwise Sip:Provider mr3.5.1 allows you to protect your VoIP system against SIP attacks, in particular Denial of Service and brute-force attacks.
Let’s go through each of those attacks and let´s see how to configure your system in order to face such situations and react against them.
We are going see, as well, how increase your security and how it´s easy to integrate fail2ban with your NGCP system, in order to ban attackers IPs.

Denial of Service

As soon as you have packets arriving on your NGCP server, it will require a bit of time of your CPU. Denial of Service attacks are aimed to break down your system by sending floods of SIP messages in a very short period of time and keep your system busy to handle such huge amout of requests.
NGCP allow you to block such kind of attack quite easily, by configuring the following section in your /etc/ngcp-config/config.yml:

 

security:
   dos_ban_enable: 'yes'
   dos_ban_time: 3600
   dos_reqs_density_per_unit: 50
   dos_sampling_time_unit: 2

 

Basically, as soon as NGCP receives more than 50 messages from the same IP in a time window of 2 seconds, that IP will be block for 3600 sec, and you will see in the the kamailio-lb.log a line saying:

 

Nov 9 00:11:53 sp1 lb[41958]: WARNING: <script>: IP '1.2.3.4' is blocked and banned - R=<null> ID=304153-3624477113-19168@tedadg.testlab.local

 

The banned IP will be stored in kamailio memory, you can check the list via web interface or via the following command:
 
# ngcp-kamctl lb fifo sht_dump ipban

Bruteforcing SIP credentials

This is a very common attack you can easily detect checking your /var/log/ngcp/kamailio-proxy.log. You will see INVITE/REGISTER messages coming in with strange username. Attackers is trying to spoof/guess subscriber´s credentials, which allow them to call out.
The very first protection against these attacks is : use STRONG passwords. Always.
Nevertheless NGCP allow you to detect and block such attacks quite easily, by configuring the following /etc/ngcp-config/config.yml section :
 
  failed_auth_attempts: 3
  failed_auth_ban_enable: 'yes'
  failed_auth_ban_time: 3600

 

You may increase the number of failed attempt if you want (in same cases it´s better to be safed, some users can be banned accidentally because they are not writing the right password) and adjust the ban time. If a user try to authenticate an INVITE (or REGISTER) for example and it fails more then 3 times, the “user@domain” (not the IP as for Denial of Service attack) will be block for 3600 seconds.
In this case you will see in your kamailio-lb.log the following lines:

 

Nov 9 13:31:56 sp1 lb[41952]: WARNING: <script>: Consecutive Authentication Failure for 'dgrotti@mydomain.com' UA='Linphone' IP='192.168.0.3' - R=<null> ID=313793-3624525116-589163@testlab.local
 
Both the banned IPs and banned users are shown in the Admin web interface, you can check them by accessing the “Security Bans” section in the main menu.
You can check the banned user as well by retrieving the same info directly from kamailio memory, using the following commands:
 
# ngcp-kamctl lb fifo sht_dump auth

 
Additionally you can check the UA value from the log line, and decide to add that User Agent to your User Agent blacklist (see “Blocking User Agent” paragraph).

Increasing your security

Let’s see now how to increase your NGCP security by adding an additional security check and how to integrate that with Fail2ban.
Malicious attacker usually use well-know tools (e.g. sipvicious) and well known User Agents. An additional level of security would be blocking IPs based on the SIP User Agent, let’s see the step to implement that into you NGCP and how to integrate this with fail2ban.

 

Blocking User Agent

In order to mitigate attack and malicious users based on SIP UA you need to patch your kamailio configuration. If you want to block his IP you have to install and configure fail2ban.

Let’s start to create our customtt file:

 

cp /etc/ngcp-config/templates/etc/kamailio/lb/kamailio.tt2 /etc/ngcp-config/templates/etc/kamailio/lb/kamailio.customtt.tt2

 

open you /etc/ngcp-config/templates/etc/kamailio/lb/kamailio.customtt.tt2 file and add the following lines:

 

route
{
...

if(!sanity_check(“1511″, “7″))
{
xlog(“L_WARN”, “Malformed SIP message detected – [% logreq_init -%]\n”);
exit;
}

## filtering by UA : blacklist
if( is_method(“REGISTER|INVITE”) && ($ua =~ “friendly-scanner” || $ua =~ “sipvicious” || $ua =~ “^sipcli.+”) )
{
            xlog(“L_WARN”, “Request rejected, malicious UA=’$ua’ from IP=$si – [% logreq_init -%]\n”);
            exit;
}
 
after that run “ngcpcfg apply”.
Now NGCP will discard all the requests coming from those malicious UAs.
But you want more! You want to block their IPs using NGCP firewall. To do that let’s see how to install and configure fail2ban to work with your NGCP.

 

Integrating fail2ban with NGCP

Just install the debian package, configuration directory will be in /etc/fail2ban:
 
# apt-get update
# apt-get install fail2ban

 
We need to modify the following file /etc/fail2ban/jail.conf. We can add IP that the system should ignore, like 127.0.0.1 and other IPs (if you have peerings for example):
 
##General section
ignoreip = 127.0.0.1 SOME OTHER NGCP IPS

 

Also we need to add to the bottom of the file the [kamailio-iptable] section:

 

[kamailio-iptables]
enabled = true
filter = kamailio
action = iptables-allports[name=KAMAILIO, protocol=all]
logpath = /var/log/ngcp/kamailio-lb.log
maxretry = 1
bantime = 3600

 

Then we need to create the filter, just creating the file /etc/fail2ban/filter.d/kamailio.conf :

 

[Definition] 
# filter for kamailio messages
failregex = Request rejected, malicious UA='.*' from IP='<HOST>'

 
In this way as soon as fail2ban will fetch that line in kamailio-lb.log, it will put the IP in iptables and block it for 1 hour.
You may decide to ban IP instead of user in case of failed authentication. This is quite easy, just change your kamailio.conf filter into:
 
[Definition]
# filter for kamailio messages
failregex = Request rejected, malicious UA='.*' from IP='<HOST>
            Consecutive Authentication Failure for '.*' UA='.*' IP='<HOST>'

 
Also you should adjust you config.yml configuration, in order to ban the user just few seconds:
 
failed_auth_attempts: 3
failed_auth_ban_enable: 'yes'
failed_auth_ban_time: 1

 
then run:
 
# ngcpcfp-apply
# /etc/init.d/fail2ban restart

 
In this way, after 3 attempts, the IP and not the user will be banned for 3600 seconds by fail2ban.
To apply the changes to fail2ban just restart the daemon:
 
# /etc/init.d/fail2ban restart
 
You can check what’s going on in /var/log/fail2ban.log (in this example ban time was 10sec):
 
2014-11-06 10:01:45,203 fail2ban.server : INFO Changed logging target to /var/log/fail2ban.log for Fail2ban v0.8.6
2014-11-06 10:01:45,206 fail2ban.jail : INFO Creating new jail 'ssh'
2014-11-06 10:01:45,206 fail2ban.jail : INFO Jail 'ssh' uses poller
2014-11-06 10:01:45,238 fail2ban.filter : INFO Added logfile = /var/log/auth.log
2014-11-06 10:01:45,240 fail2ban.filter : INFO Set maxRetry = 6
2014-11-06 10:01:45,242 fail2ban.filter : INFO Set findtime = 600
2014-11-06 10:01:45,244 fail2ban.actions: INFO Set banTime = 600
2014-11-06 10:01:45,333 fail2ban.jail : INFO Creating new jail 'kamailio-iptables'
2014-11-06 10:01:45,333 fail2ban.jail : INFO Jail 'kamailio-iptables' uses poller
2014-11-06 10:01:45,338 fail2ban.filter : INFO Added logfile = /var/log/ngcp/kamailio-lb.log
2014-11-06 10:01:45,340 fail2ban.filter : INFO Set maxRetry = 1
2014-11-06 10:01:45,342 fail2ban.filter : INFO Set findtime = 600
2014-11-06 10:01:45,343 fail2ban.actions: INFO Set banTime = 10
2014-11-06 10:01:45,370 fail2ban.jail : INFO Jail 'ssh' started
2014-11-06 10:01:45,406 fail2ban.jail : INFO Jail 'kamailio-iptables' started
2014-11-06 10:01:46,489 fail2ban.actions: WARNING [kamailio-iptables] Ban 1.1.2.12
2014-11-06 10:01:56,562 fail2ban.actions: WARNING [kamailio-iptables] Unban 1.1.2.12

 
We are working to include fail2ban in the next upcoming NGCP version.

sip:provider mr3.6.1 Released

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We are excited to announce the general availability of sip:providerCE mr3.6.1, sip:providerPRO mr3.6.1 and sip:CARRIER mr3.6.1

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 mr3.6.1?

The most important changes for mr3.6.1 are:

  • Central config management on SIP:Carrier 3.x installations
  • General system cleanup before the next LTS release
  • Full migration from Apache to Nginx
  • SOAP API is deprecated for now
  • Central storage for all SSL certificates under /etc/ngcp-config/ssl/
  • Old Subscriber interface has been disabled
  • New Subscriber interface has been separated from ngcp-panel Admin interface
  • XMPP Multi-user Chat (MUC) is enabled in Prosody
  • CloudPBX: Implement park/unpark via VSC: *97* to park and *98* to unpark the call
  • CloudPBX: Add auto-provisioning for Panasonic devices
  • New REST API methods are now available (e.g. search for customers by external_id, resellers can manage callforwards and faxserver settings, add credits via REST API and more)
  • CDR-exporter: dynamic field selection and sftp transfer
  • Custom fields gpp0-gpp9 can be stored in CDR tables
  • SSLv3 is completely disabled (CVE-2014-3566)

Important upgrade notes:

  • Apache is not going to be installed on new installations since mr3.6.1, while we keep it installed during upgrade. Feel free to remove it manually using “apt-get purge apache2″ if you do not use it. Otherwise, please migrate your virtualhosts to Nginx and remove Apache.
  • SOAP/XMLRPC API has been migrated to Nginx and disabled by default. They are deprecated for now, please switch your components to REST API. Still, SOAP/XMLRPC API are available and can be enabled in config.yml (ossbss->frontend:fcgi). They are going to be deleted in upcoming release(s).
  • Old Subscriber selfcare interface is deprecated and has been disabled by default. New ngcp-panel based Subscriber interface has been separated from ngcp-panel Admin interface and available on the old port 443. System owners can separate and control the network interfaces where Subscriber and Admin panels are available (please use web_ext type to assign Subscriber selfcare interface in network.yml and web_int for Admin interface). IMPORTANT: Admin ngcp-panel interface listens on localhost only by default, please assign “web_int” type to appropriate network interface in /etc/ngcp-config/network.yml and run ngcpcfg apply.
  • All default SSL certificates described in config.yml were copied to folder /etc/ngcp-config/ssl for central backup/management. Please check and move personal SSL certificates to new storage for further usage.

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. For those of you using Amazon Cloud we provide the EC2 AMIs in the following regions: us-east-1: ami-0033a668, us-west-2: ami-5b165c6b, us-west-1: ami-2feff96a, eu-west-1: ami-8ad661fd, ap-southeast-1: ami-8298bbd0, ap-southeast-2: ami-59294763, ap-northeast-1: ami-0b9aa20a, sa-east-1: ami-df63d5c2.

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 mr3.6.1 from scratch.

For users of the SPCE mr3.5.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.

How can I contribute to the project?

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.

Acknowledgements

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 mr3.6.1 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#10117 Fixed SOAP API under nginx (WSDL caching has been improved)
MT#10077 Fixed anonymous calls between subscribers of same customer
MT#10061 Invoice module calculates with different precisions than rate-o-mat
MT#10053 Disable sems 1.6 tm blacklist
MT#10041 REST API request with noAdministrator credentials returns error
MT#10021 PRO: Improved the ha.cf pingnodes building logic
MT#10005 w_record_route(): Double attempt to record-route
MT#9985 Enable MUC in prosody
MT#9983 PRO: Captagent process is controlled by HA now
MT#9971 PRO: ngcpcfg restarts monit first to prevent errors
MT#9965 PRO: skip heartbeat restart on update monit-services
MT#9949 Kamailio-Prox TCP_KEEPIDLE requires SO_KEEPALIVE to function (tcp idle timer)
MT#9941 API REST: PBX Group creation do not set correctly cloud_pbx_hunt_timeout
MT#9937 Carrier 3.x carrier handbook package has been created
MT#9901 CloudPBX: Implement park/unpark via VSC *97* to park and *98* to unpark
MT#9897 Customer Concurrent_max_total_out and max_total
MT#9889 Prosody: Added Audiocodes devices workaround
MT#9885 Added XMPP Last Activity to Prosody
MT#9881 CloudPBX: Implement Panasonic network phone book
MT#9869 Fix bashism on fastcgi daemons
MT#9857 ngcpcfg-api: must run on both nodes
MT#9849 Carrier 3.x mysql: add missing auto_increment_increment option
MT#9843 Add help option to ngcp-ppa command
MT#9837 VMs PRO-trunk failed to be install due to missed web_int on eth0 of sp2
MT#9829 PRO: Can’t uninstall ngcp-templates-pro-hylafaxplus-iax on PRO 3.4+
MT#9787 Emergency call should be always reachable even with concurrent max call set
MT#9775 Carrier 3.x Fixed ngcp-upgrade to installs ngcp-ngcp-carrier package
MT#9773 Collective-check issues in trunk
MT#9745 Error in log file from ROUTE_PREPARE_MOH
MT#9733 Call get stuck if caller is a PBX user and caller max out is reached
MT#9711 CloudPBX: Implement directed call pickup
MT#9709 Fixed error: dispatcher [dispatch.c:1240]: ds_get_index(): destination set [50] not found
MT#9677 Disable old ngcp-panel selfcare interface and move new one to port 443
MT#9675 Remove old ngcp components: subscriber selfcare and apache2
MT#9671 REST API: Search for customers by external_id
MT#9653 API: allow resellers to access callforwards and faxserver settings
MT#9643 Fixed deadlock in tm module
MT#9637 cdr-exporter: implement dynamic field selection and sftp transfer
MT#9633 Sems in prepaid scenario seems to check the wrong database
MT#9623 ngcp-status: ignore /usr/share/keyrings/debian-archive-removed-keys.gpg failure
MT#9607 Failed to terminate subscriber
MT#9603 Fixed lintian errors on captagent package
MT#9601 Rewrite rules are not applied on blind transfer
MT#9599 Fails to delete Allowed_ips list
MT#9597 [Security issue] Disable SSLv3 due to “Poodle (CVE-2014-3566)”
MT#9583 Carrier 3.x MySQL on proxy nodes occupied 100% RAM for bufferpool
MT#9579 Carrier 3.x database.central.dbhost must be switched from db01a to db01
MT#9573 Fixed MySQL perms for sp1/sp2
MT#9569 Empty user in location table generate invalid INVITE for SEMS
MT#9567 Port 1443 was reachable on every IPs (binding to web_ext/web_int now)
MT#9565 Define gpp0-gpp9 as usr-prefs and pass them to CDRs
MT#9563 Office hours not triggered for huntgroup
MT#9561 kamailio crash on dialog timer in mr3.5
MT#9547 ngcp-ossbss migration to nginx
MT#9515 sems-pbx reply 488 Invalid Fingerprint in mr3.4.2 to call to auto-attendand
MT#9497 Remove automatic build of hylafax configs by ngcp-templates-*-hylafax package
MT#9481 REST API: new method to add credits
MT#9479 NGCP related cron jobs are being installed and executed on all the nodes
MT#9447 REST API: Search for customers by external_id
MT#9441 Call Forwarding to external Voice-Mail doesn’t work anymore
MT#9439 Multiple rewrites of Allow header produce malformed heaeder
MT#9431 Huntgroup policy provisioned via API seems not properly applied in DB
MT#9423 Improve ngcp-ppa usability
MT#9385 Carrier 3.x obtain a unique dbnode
MT#9287 Improve ngcpcfg to handle Carrier 3.x configs from central place (mgmt node)
MT#9285 Redesign config.yml to be equal on all hosts.
MT#9281 Redesign network.yml to be equel on all hosts.
MT#9277 PRO: IVR Voucher code to add credit
MT#9275 PRO: IVR check the credit status
MT#9209 Introduce preference allowed_clis_reject_policy
MT#9171 Updated package captagent
MT#9155 Carrier 3.x Fixed LB error:  lval_assign (): assignment failed at pos (588,22-588,56)
MT#8995 Removed hardcoded “sp1″ and “sp2″ from snmpd.conf.tt2
MT#8961 auth.log contains tons of “pam_env(cron:session): Unable to open env file: /etc/default/locale: No such file or directory”
MT#8895 Moving hylafax and iaxmodem under monit
MT#8773 ngcpcfg-api: adapt api to 3.x
MT#8759 Carrier 3.x Aliases for connecting to specific mysql instance
MT#8741 Remove unnecessary [- IF PRO -] checks in case of using ngcp-check_active
MT#8661 Rename mediaproxy-ng to rtpengine in public documentation
MT#8311 Carrier 3.x ngcp-collective-check support
MT#8299 CloudPBX: implement visual key/line selection in web interface
MT#8245 Separate CSC and admin web interface from network perspective
MT#8117 Introduce: role db
MT#6969 ngcp-panel: trigger SPA device resync via SIP NOTIFY
MT#6499 kamailio-config-tests: fix problems on EC2 environment
MT#6243 allow multipart body on kamailio

sip:provider mr3.6.2 Released

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We are excited to announce the general availability of sip:providerCE mr3.6.2 and sip:providerPRO mr3.6.2.

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 mr3.6.2?

The build mr3.6.2 contains the set of fixes for release mr3.6. No new features added to mr3.6.2. See the full list of changes at the end of the announce.

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-a07710c8
  • AMI ID for region us-west-2: ami-53f0a663
  • AMI ID for region us-west-1: ami-7b5e4c3e
  • AMI ID for region eu-west-1: ami-a2279bd5
  • AMI ID for region eu-central-1: ami-da91a0c7
  • AMI ID for region ap-southeast-1: ami-61cbe733
  • AMI ID for region ap-southeast-2: ami-2f7b1315
  • AMI ID for region ap-northeast-1: ami-fa111cfb
  • AMI ID for region sa-east-1: ami-2369d93e

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 mr3.6.2 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.

What’s coming up next?

The mr3.6.2 build is stabilization/bugfixes build, so contains no new features. We are preparing some exciting new features for the upcoming mr3.7.1. Also, there will be some further enhancements to the REST-API, so any feedback on the current state is highly appreciated. The SOAP/XMLRPC APIs have been migrated to Nginx and disabled by default in the previous build mr3.6.1. You may switch it on using config.yml setting ossbss->frontend:fcgi, but those obsolete APIs are going to be deleted in the upcoming release(s).

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 mr3.6.2 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#9949 compile fix for the tcp idle timer in db_postgres kamailio module
MT#9763 configure the NGCP push notification for chat (XMPP)
MT#9643 safety catch inside t_check_trans() of tm module (fixes deadlock e.g. on 302)
MT#9637 cdr-exporter: implement dynamic field selection and sftp transfer
MT#9603 lintian errors on captagent package
MT#9567 ngcp-panel: listen on all ips not only the shared ones; Add web_int handling to ngcp-eaddress
MT#9547 ngcp-ossbss migration to nginx: improved access perms, don’t try to generate ssl certs if it’s already there
MT#9287 Improve ngcpcfg to handle Carrier 3.x configs from central place (mgmt node)
MT#8613 API: add LI functions to REST
MT#7793 cdr/event-exporters: fix alias handling and merge updates
MT#10421 Prepare and release mr3.6.2
MT#10381 add concurrent_max and concurrent_max_out contract preferences
MT#10363 ngcp-collective-check complains on missed apache2 on management node
MT#10331 Carrier 3.x NGCP elasticsearch cron cleanup cannot connect elasticsearch server
MT#10309 Cannot set password_min_length in Panel to less than 6.
MT#10273 Fixed NPN setting on call from Faxserver
MT#10257 SPCE Installation from ISO fails (ngcp-asterisk failed to install on CPU without PrefetchW instruction
MT#10213 Check content of /etc/ngcp_nodename before upgrade (CE should use ‘spce’ in DB also)
MT#10135 [CloudPBX] Disregard CFNA for Subscriber when called within a HuntGroup (fixed loop)
MT#9941 [CloudPBX] Add cloud_pbx_hunt_* fields to rest api.
MT#10319 [CloudPBX] fixed Call Forward from the hunt group
MT#10077 [CloudPBX] “Anonymous call rejection for call between subscribers of same customer
MT#10117 soap api doesn’t work under nginx
MT#10093 be able to install the same set of packages on CARRIER nodes
MT#10065 ngcpcf-api: network interfaces order
MT#10045 unable to uncheck the box saying “Charge VAT”
MT#10041 REST API: can’t get subscriberregistrations as non-administrator
MT#10355 REST API: fixed clir preference inserted into DB as empty string
MT#10021 [PRO] Change the ha.cf pingnodes building

sip:provider mr3.7.1 Released

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We are excited to announce the general availability of sip:providerCE mr3.7.1 and sip:providerPRO mr3.7.1.

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 mr3.7.1?

The most important changes for mr3.7.1 are:

  • Call queues module for Cloud PBX
  • Panasonic and Yealink IP phones auto-provisioning
  • REST API: add filter for api/calls
  • Provide pre-recorded ngcp-prompts in multiple languages and a preference to select the language
  • Support separate NCOS type for Call Forwarding
  • Implement To header manipulations using outbound_to_user preference
  • New modes for handling the extension dialed behind main number using preference extended_dialing_mode
  • Improved sip:carrier system deployment using PXE boot and apt proxy cache
  • Improved sip:carrier system management using klish and SNMP monitoring
  • Improved collection of call counters in redis
  • Backported new Linux kernel 3.16+62~bpo70+1 as a part of mr3.8 LTS preparation

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-04611f6c
  • AMI ID for region us-west-2: ami-7b0d534b
  • AMI ID for region us-west-1: ami-439a8406
  • AMI ID for region eu-west-1: ami-8bef6cfc
  • AMI ID for region eu-central-1: ami-cc5b68d1
  • AMI ID for region ap-southeast-1: ami-c08ba392
  • AMI ID for region ap-southeast-2: ami-97483dad
  • AMI ID for region ap-northeast-1: ami-0e08150f
  • AMI ID for region sa-east-1: ami-dd4ff2c0

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 mr3.7.1 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 mr3.7.1 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#9985 “Enable MUC in prosody”
MT#9983 “Captagent process has to be controlled by HA”
MT#9971 “ngcpcfg restarts services in random order which cause errors”
MT#9967 “proxy uses wrong redis db number for storing the dialogs”
MT#9965 “Skip heartbeat restart on update monit-services”
MT#9941 “API REST: PBX Group creation do not set correctly cloud_pbx_hunt_timeout”
MT#9937 “missing carrier handbook package”
MT#9901 “CloudPBX: Implement park/unpark via VSC”
MT#9897 “Customer Concurrent_max_total_out and max_total”
MT#9885 “XMPP Last Activity”
MT#9881 “CloudPBX: Implement Panasonic network phone book”
MT#9811 “mysqlbinlog fails to execute: unknown variable ‘default_character_set=utf8′”
MT#9803 “registered_users and registered_devices statistic in SNMP on Carrier shows wrong value”
MT#9801 “provisioned_subscribers statistic in SNMP on Carrier shows wrong value”
MT#9675 “Remove old ngcp components: subscriber selfcare and apache2″
MT#9643 “deadlock in tm module”
MT#9637 “cdr-exporter: implement dynamic field selection and sftp transfer”
MT#9617 “heartbeat-2 fails to build against Debian/jessie”
MT#9603 “lintian errors on captagent package”
MT#9585 “rtpengine kernel module still has name xt_MEDIAPROXY”
MT#9567 “port 1443 is reachable on every IPs”
MT#9547 “ngcp-ossbss migration to nginx”
MT#9497 “remove automatic build of hylafax configs by ngcp-templates-*-hylafax package”
MT#9453 “[Netcologne] New feature request in order to disable some call forward” “reopened”
MT#9287 “Improve ngcpcfg to handle Carrier 3.x configs from central place (mgmt node)”
MT#9285 “Redesign config.yml to be equel on all hosts.”
MT#9261 “iaxmodem fails to build against Debian/jessie”
MT#9259 “hylafaxplus fails to compile against Debian/jessie”
MT#9257 “kamailio: fails to compile against Debian/jessie because libradiusclient-ng-dev doesn’t exist”
MT#9177 “Implement authentication for device provisioning from web”
MT#9171 “Package captagent for Debian” “reopened”
MT#9155 “[Carrier 3.x] LB error: lval_assign (): assignment failed at pos (588,22-588,56)”
MT#8645 “Allow to terminate billing profiles to not show in interface anymore”
MT#8299 “CloudPBX: implement visual key/line selection in web interface”
MT#8173 “Stop NGCP/ALL services before upgrade”
MT#8129 “PIN code for entrance to conference room”
MT#7999 “Update linux-image-amd64 to recent wheezy-backports version”
MT#7793 “Implement event billing”
MT#7705 “Increase the oss.log. No significant info provided right now”
MT#7153 “Missing “Create PBX Group” button if not defined the max extension number in the customer”
MT#6785 “can’t create a PBX group with extension 0″
MT#6773 “subscsriber preferences API call doesn’t return the collection”
MT#5715 “heartbeat: sometimes cannot be restarted due to error in log”
MT#5241 “ngcp-panel: can’t properly edit extension of pbx group”
MT#10869 “Disable certain Jitsi autoprov features”
MT#10843 “CE upgrade claims on duplicate revisions in cfg/db_schema tables”
MT#10797 “Absent mandatory parameter “type” for customer creation makes 500 Server response”
MT#10789 “User-Agent: inserted in case of CFU”
MT#10785 “document how to skip mysql replication on installer”
MT#10769 “no mysql replication from sp1 to sp2 on new installed systems”
MT#10761 “update sems dsm scripts for new ngcp-prompts”
MT#10749 “dlgcounter total gets increased twice in a local call”
MT#10733 “haproxy can’t log to syslog”
MT#10717 “LI: upgrade captagent package for bugfixing”
MT#10711 “installer: use file to pass arguments/options”
MT#10709 “kamailio dlgcnt.lua template: wrong implementation of set_dlg_profile”
MT#10705 “CloudPBX: Allow to set group members from within group”
MT#10703 “script for configure MySQL replication between sp1/sp2 nodes”
MT#10701 “installer: allow skip Mysql replication”
MT#10699 “ngcpcfg: use sqlite for manage cfg_schema”
MT#10695 “ngcp-sync-constants: create expecific option to check and set master replication”
MT#10691 “integrate call queue module for pbx”
MT#10689 “Heartbeat creates a lock file in tmp folder which is not secure, please move it to /var/run/heartbeat/”
MT#10659 “check-for-mysql: add parameter to check second instance”
MT#10649 “/etc/cron.d/voisniff runs on all the nodes of Carrier 3.x setup even if voisniff is disabled”
MT#10639 “sync errors in db01X nodes after installation”
MT#10635 “Carrier 3.x: dispatcher is missing the LB IP when cluster_sets.type=’central'” “reopened”
MT#10621 “ignore CLIR when Auto Attendant forwards a call”
MT#10619 “Add option to start/not start hb_watchdog to config.yml”
MT#10607 “Monit’s protocol check for exim check randomly fails although exim is running and port 25 reachable”
MT#10603 “Rate-limit autoprov access”
MT#10601 “dlg counters: keys in central redis are not removed”
MT#10591 “rtpengine_sock contains both sip_int and rtp_int on Carrier”
MT#10585 “add rtpengine control port to config.yml”
MT#10581 “motd not in etckeeper gitignore file” “reopened”
MT#10577 “Upgrade mr3.6->mr3.7 requires upgrade trick for ngcp-prompts package installation”
MT#10571 “ngcpcfg-api: error when no /var/run/fastcgi dir exists”
MT#10565 “corner cases where ROUTE_SET_CALLER_DIALOG is invoked”
MT#10555 “ngcp-sems and sems-pbx should listen on sip_int for XMLRPC”
MT#10551 “Rate-o-mat regularly crashing on carrier due to MySQL deadlock situation” “reopened”
MT#10531 “need ngcp-asterisk-sounds on CE only in 3.7″
MT#10525 “Monit config should respect the load thresholds from config.yml”
MT#10521 “Get rid of “Failed to decrease redis counters” proxy errors in case of call forwards”
MT#10509 “installer: config.yml.factory_defaults changes gets commited on all nodes from central mgmt”
MT#10493 “ngcpcfg tt2-daemon should pass copy of YML object structure to “$tt->process””
MT#10491 “prosody: use libevent to improve connection monitoring”
MT#10421 “Prepare and release mr3.6.2″
MT#10419 “CLIP in case of Call Forwarding”
MT#10383 “Monit sometimes reports “proxy Execution failed” for normally running processes”
MT#10381 “add concurrent_max and concurrent_max_out contract preferences”
MT#10379 “add dnsmasq to monit”
MT#10377 “[App] When wifi is disable and app goes offline, the other user see the buddy still online”
MT#10371 “Two mysql instances are running on LB node Carrier 3.x mr3.6.1 after update to the last hotfix”
MT#10367 “Make language for voicemail/conference sounds a user preference”
MT#10363 “ngcp-collective-check claims on missed apache2 on management node”
MT#10355 “Required by NGCP::Panel::Role::API::Preferences::check_pref_value (JSON::is_bool)(false) inserted into DB as empty string”
MT#10341 “Finalize panasonic provisioning integration”
MT#10337 “Add update/override to cloudpbx-devices tool”
MT#10331 “Carrier 3.x NGCP elasticsearch cron cleanup cannot connect elasticsearch server”
MT#10319 “Call Forward from the hunt group”
MT#10303 “Unsync between location DB and kamailio memory when REGISTER has different CallID”
MT#10273 “Webfax problem in combination with hardware fax solution”
MT#10263 “provide debug package for ngcp-asterisk”
MT#10261 “ngcp-asterisk package lacks conflicts with asterisk package”
MT#10257 “SPCE Installation from ISO fails (ngcp-asterisk failed to install on CPU without PrefetchW instruction)”
MT#10213 “Check content of /etc/ngcp_nodename before upgrade (CE should use ‘spce’ in DB also)”
MT#10199 “Add NCOS Level in Subscriber Profile”
MT#10169 “Allow peering rules, rewrite rules and peerings to be individually disabled”
MT#10145 “event-exporter failis to execute: “Can’t use string (“{ accounting.events.export_statu”…) as an ARRAY ref …””
MT#10135 “Disregard CFs for Subscriber when called within a HuntGroup”
MT#10117 “soap api doesn’t work under nginx”
MT#10079 “heartbeat udpport conflicts with nfs.rpcbind service”
MT#10065 “ngcpcf-api: network interefaces order”
MT#10061 “Invoice module calculates with different precisions than rate-o-mat”
MT#10057 “Add config.yml description to handbook”
MT#10053 “disable sems 1.6 tm blacklist”
MT#10425 “Carrier 3.x Improve system management using clish”
MT#10041 “REST API request with non “Administrator” credentials returns error”
MT#10029 “implement To header manipulations”
MT#10025 “New modes for handling the extension dialed behind main number”
MT#10021 “Change the ha.cf pingnodes building”
MT#10005 “w_record_route(): Double attempt to record-route”
MT#10181 “default SSL setup not accessible via Chrome + Chromium”
MT#10309 “Cannot set password_min_length in Panel to less than 6.”
MT#10343 “Add hunting indicator in display name if call to device during hunting”
MT#10301 “CloudPBX pilot subscriber looses Administrator flag – not possible to set it again”
MT#10587 “Implement Yealink auto-provisioning”
MT#10543 “Control T38 for auto-provisioning via usr preference”
MT#10537 “Implement calllists as simplified way to fetch call history for a subscriber”
MT#10501 “Devices page too slowly”

Tips&Tricks: How to use Serial Forking based on q-value with Sip:Provider

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Sipwise Sip:Provider (aka NGCP) allows you to register multiple devices under the same subscriber. By the default the maximum number of device you can register is 5, and this value is configurable via config.yml.
If a customer registers multiple devices, NGCP – once receives a call for that user – just send the call to all the registered devices, in parallel. All the devices will ring at the same time. This is called Parallel Forking, and this is the default behavior. But what NGCP can also do is the so-called Serial Forking, which mean let ring one device first, then after a timeout let ring the next device, and so on and so forth. The Serial Forking feature has been introduce since version mr3.3.1, and it could be quite interesting to have it, but maybe not all of the NGCP users know how it works.

Serial Forking

Serial Forking just calls the registered devices in serial, one after the anothers. But how ? How can I tell the system which one I want to call as first, and which one as second ?
Serial Forking is based on SIP Contact’s parameter called q-value, which is basically a priority number, set by the clients during their Registration.
The q value is a floating point number in a range 0 to 1.0 specify as parameter in the Contact header field. The higher the q value number, the more priority that device has. Contacts with q value 1.0 have maximum priority, so such contacts will be always tried first in serial forking. Contacts with q value 0 have the lowest priority and they will be tried after all other contacts with higher priority.
Here an example of Contact with q value set to 0.9:

Contact: <sip:johdoe@sip.domain.com:5060;transport=udp>;expires=3600;q=0.9

So, q-value is set by the client during the Registration process.
In most of the cases clients don’t set q value and do not allow you to set it, so NGCP just set a default value of ‘q=-1′ in the database, which means basically it’s not going to use it and it’s not going to perform any serial forking.
As you may understood, there is a big problem at this point.

What’s the problem with q-value ?

The majority of SIP Phones out there don’t allow you to set q-value!
Which means, you can NOT use serial forking in NGCP!
This is pity, but frankly there is nothing NGCP can do to avoid that, this is a Client feature, and SIP clients don’t have it.

How to use Serial Forking in NGCP

But with some tricks, we can handle this, anyway…
What we are suggesting here is a way to set q-value on your client and use Serial Forking.
Clients don’t allow us to set q-value at all, but there is another header that almost all clients allow you to set, and this is the User-Agent header field.
So the idea is to set the q-value in the User-Agent header field of your phones, then parse it on NGCP and use it as real q-value for your Serial Forking! Let’s see how to do it.

Setting q-value in User-Agent header

I will take as example the SIP Phone CISCO SPA502g. But this is possible with all the CISCO SPA50X and with other brands.
If you access the phone web interface, just click on ‘Admin Login’ and then ‘Advanced’ on the top-right.
Now go into ‘SIP’ section and look for the field ‘SIP Reg User Agent Name:‘.
Just set there the following string, ‘$VERSION;q=0.9‘. $VERSION is the internal variable used by default as standard User Agent name, and it looks like ‘Cisco/SPA502G-7.4.9c’, but with our additional parameter it will look like ‘Cisco/SPA502G-7.4.9c;q=0.9‘.

Handling q-value on NCGCP

Now, on NGCP we are going to add the following section in our /etc/ngcp-config/templates/etc/kamailio/lb/kamailio.cfg.customtt.tt2 file:


xlog("L_NOTICE", "New request - M=$rm R=$ru F=$fu T=$tu IP=$pr:$si:$sp ID=$ci\n");
setflag(FLAG_OUTBOUND);
}


### Patch ### Read q-value from User-Agent header and use it.
if(is_method("REGISTER") && is_present_hf("Contact"))
{
$var(qval)=$(hdr(User-Agent){param.value,q});
if($var(qval) != $null)
{
xlog("L_NOTICE", "q-value set by the client is q=$var(qval) - R=$ru ID=$ci\n");
$var(newct) = $ct + ';q=' + $var(qval);
remove_hf("Contact");
append_hf("Contact: $(var(newct))\r\n");
}
}
### end Patch ###

then apply the changes.
This part will read the q value in the User-Agent header – if exist – and append the value in the contact header.
So, for example, subscriber John Doe has two phones, CISCO phone 1 with q=0.9 in his User-Agent header (source IP 1.1.1.1), and CISCO phone 2 with set q=0.1 (source IP 2.2.2.2).
John will register both devices, and the situation will be the following:


mysql> select username,domain,contact,q from kamailio.location where username='johndoe';
+----------+-------------+-------------------------------+-------+
| username | domain      | contact                       | q     |
+----------+-------------+-------------------------------+-------+
| johndoe  | sip.dom.com | sip:johndoe@1.1.1.1:5060      | 0.90  |
| johndoe  | sip.dom.com | sip:johndoe@2.2.2.2:5060      | 0.10  |
+----------+-------------+-------------------------------+-------+

Now, in order to enable Serial Forking for subscriber John Doe, we just need to go into Subscriber’s Preferences, section ‘Internals’ and enable “serial_forking_by_q_value”.
Every calls – now – to John Doe will let ring CISCO Phone 1 first, and then phone 2, in serial.

Conclusion

If you are able to set a new SIP header field on your client, it would be much better, so you can use a dedicated header for that, example “X-q-value: q-value header;q=xx”.
You just need to parse header ‘X-q-value’ insteado of ‘User-Agent’ in the kamailio patch above.
Also, another solution could be to set a static registration (if you know the devices IPs) and then set the q-value manually in the DB and then restart kamailio proxy to load the change on kamailio memory as well.

Assistent/in der Geschäftsführung (w/m)

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Sipwise ist ein junges österreichisches Unternehmen in der IT / Telekommunikationsbranche. Unsere auf
Open-Source-Technologie basierenden Produkte werden weltweit von Netzbetreibern erfolgreich eingesetzt.

 

Wir sind ein engagiertes Team, im Begriff, den Telekommunikationsmarkt nachhaltig zu prägen und suchen
ab sofort Unterstützung unseres Teams am Campus21 bei Wien:

 

Assistent/in der Geschäftsführung (w/m)

Als Assistent(in) der Geschäftsführung übernehmen Sie selbständig sämtliche Office Agenden zur Entlastung
der Geschäftsführer und unterstützen das gesamte Team in organisatorischen sowie administrativen Tätigkeiten.
Dazu zählt auch die eigenverantwortliche Übernahme diverser Projekte.

 

Wir wenden uns an dienstleistungsorientierte Persönlichkeiten mit strukturierter Arbeitsweise und Organisationsstärke.

 

Ihre Aufgaben im Detail
Office Management: Bestellungen, Recherchetätigkeiten (Angebote einholen, vergleichen und aufbereiten),
  Erarbeitung und Aufbereitung von Listen und sonstigen Informationsdokumenten, Betreuung & Bewirtung unserer
  internationalen Gäste, Schriftverkehr und Terminvereinbarung (dt/en)
Travel Management (Flug- und Hotelbuchungen)
> Spesen- und Reisekostenabrechnungen (Erstellung, Kontrolle Belege)
> Sales Support: Unterstützung des Vertriebsteams in administrativen und organisatorischen Belangen, Sales Support
  (CRM aktualisieren, Visitenkarten eintragen, Versand Weihnachtsgeschenke, etc.)
> Bürowege, Telefondienst (dt/en)

 

Ihr Profil
> selbständige und eigenverantwortliche Arbeitsweise, zuverlässige und kooperative Persönlichkeit
> sehr gute EDV-Kenntnisse (MS Office)
> sehr gute Englischkenntnisse in Wort und Schrift (Unternehmenssprache Englisch)
> kaufmännische Ausbildung von Vorteil (FH, HAK, HASCH, Lehre)

 

Unser Angebot
> Mitarbeit in einem internationalen Team
> abwechslungsreiches Aufgabengebiet und die Möglichkeit, Verantwortung für Teilbereiche zu übernehmen
  und selbständig zu arbeiten
> Vollzeitbeschäftigung im Ausmaß von 38,5 Wochenstunden mit Gleitzeitvereinbarung
> abhängig von Ihrer Qualifikation und Erfahrung bieten wir eine marktkonforme Überzahlung des
  KV-Mindestgrundgehaltes von EUR 2.188,00 brutto monatlich.

 

Es erwartet Sie eine abwechslungsreiche Aufgabe in einem international wachsenden Unternehmen und einem dynamischen Marktumfeld!
Wir freuen uns auf Ihre aussagekräftige Bewerbung: jobs@sipwise.com
Sipwise GmbH, Europaring F15, 2345 Brunn am Gebirge

Sipwise in Cape Town

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Sipwise at the AFRICAcom in Cape Town in November this year

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.

floorplan incl Sipwise

 

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


Sip:Provider as a Cloud PBX Solution

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A cloud PBX is a PBX serving an office or a company, not located inside them but somewhere out there. It’s delivered as a hosted service, typically by telephone companies. No hardware for end customers, only phones.

With the Sipwise’s Sip:Provider PRO, a VoIP ISP can easily provide such feature to his end customers, in particular offices and small companies.The Cloud PBX module is a commercial module you can add on top of your basic Sip:Provider PRO in order to host cloud PBXs with basic functionalities such as IVR, Announcements, Voiceboxes, Call Group, Queues, Phone Auto-Provisioning system, Chat (XMPP), Call Transfers, BLF and Shared Lines support, Call Pick-up.

The ISP just need to create a user the end customer can use to access the web interface and to manage his own cloud PBX directly. He will be able to create extensions, groups, chat buddy list, queue and provision his phones.

Let’s see briefly how it works.

 

Creating the PBX

The ISP has just to create one special subscriber called Pilot Subscriber, specifying some mandatory values such as the main number associated to the PBX (e.g 431555123, so each 3 digits extension will look like 431555123XXX), the maximum number of extensions can be created and the web credentials for the Pilot Subscriber. That’s all.

 

Managing the PBX

The office/company IT manager, or the person in charge to configure the PBX, needs to access the web interface (https://mypbx.domain.com) with the Pilot Subscriber’s credentials and just start to create the extensions number and associates them to each phones:

 

 

We support a wide range of brands, such as Cisco/Linksys, Polycom, Yealink, Panasonic, and much more will come.

Once the association is finished, he just need to push and sync the configuration directly to the phone, specifying the phone’s IP address.
Then he can start to create Call Groups, IVR and office announcements and set the office sounds like music on hold or Queue messages. All the phones are now configured and ready to be used. Additionally, for all the PBX extensions, the system will automatically create the chat users, so every employees can see each other and talk together via an XMPP client.
To find out more about the Cloud PBX solution, feel free to contact us!

sip:provider mr3.7.2 Released

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We are excited to announce the general availability of sip:providerCE mr3.7.2 and sip:providerPRO mr3.7.2.

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 mr3.7.2?

The build mr3.7.2 contains the set of fixes for release mr3.7. No new features added to mr3.7.2. See the full list of changes at the end of the announce.

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-be451fd6
  • AMI ID for region us-west-2: ami-e3d3f0d3
  • AMI ID for region us-west-1: ami-cf59bd8b
  • AMI ID for region eu-central-1: ami-4ead9f53
  • AMI ID for region eu-west-1: ami-9754c7e0
  • AMI ID for region ap-southeast-1: ami-de487e8c
  • AMI ID for region ap-southeast-2: ami-c1a7d6fb
  • AMI ID for region ap-northeast-1: ami-670ae367
  • AMI ID for region sa-east-1: ami-cf3d84d2

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 mr3.7.2 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.

What’s coming up next?

The mr3.7.2 build is stabilization/bugfixes build, so contains no new features. We are preparing some exciting new features for the upcoming mr3.8.1. Also, there will be some further enhancements to the REST-API, so any feedback on the current state is highly appreciated. The SOAP/XMLRPC APIs have been migrated to Nginx and disabled by default in the previous build mr3.7.1. You may switch it on using config.yml setting ossbss->frontend:fcgi, but those obsolete APIs are going to be deleted in the upcoming release(s).

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 mr3.7.2 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#11947 Error validating sipwise keyring on CE install CD
MT#11891 Rewrite rule’s priority change does not trigger xmlrpc request
MT#11879 Prosody console still listen on loopback even if dummy0 with sip_int is specified
MT#11753 BLF NOTIFY is sent without body in mr3.7.1
MT#11741 Add API function to download device front/mac images
MT#11737 Fraud-lock script fails to find voip contract
MT#11733 Wrong ngcp-event-exporter cron configuration
MT#11719 Looks like we start hylafax before iaxmodem during the takeover
MT#11659 [PBX] sems change the sound cache path to be consistent with audio_cache module
MT#11651 Issues with CFT for cloud PBX subscribers
MT#11647 Wrong content-type in response header for panasonic config
MT#11595 ossbss/mysql_values.cfg.services: SQL error Column count doesn’t match value count
MT#11569 Uploading new MOH in a soundset still plays the old one
MT#11549 Search and find subscriber by Alias number
MT#11523 CloudPBX: Add Cisco 504G with one and two 500S consoles
MT#11521 Call drops when selecting AA slot that is not configured
MT#11517 API REST: PBX Hunt-Group creation/update do not set correctly cloud_pbx_hunt_timeout
MT#11515 Asterisk voicemail.conf not filled correctly by ngcpcfg apply
MT#11499 Reminder call does not work in mr3.7.1
MT#11467 P-Callee-UUID is ‘0’ in case of circular ringing
MT#11377 Implement fallback to system sound set for callqueue
MT#11371 /usr/lib/ngcp-ngcpcfg/get_default_pingnodes grepping commented nameserver
MT#11321 Problem with “Peering Costs” and source_carrier_cost
MT#11299 Wrong P-App-Params on call from PBX to AA/OH
MT#11295 Adding Session-Expires header to the SEMS whitelist by default
MT#11285 500 http response code in the panel in Device->dev_field_config
MT#11279 Add config option to skip device bootstrap vendor RPC calls
MT#11243 [Carrier] Add ability to completle disable PXE boot ( dnsmasq )
MT#11241 Several improvements for confpin.dsm
MT#11239 dsm/mod_utils: rename the prompts for single digits after the tens
MT#11237 [PBX] Fix MWI on for removed line
MT#11233 extra_socket currently broken in mr3.7.1
MT#11219 Load default sound files from file system and provide fallback from customer to system
MT#11125 ngcp-prosody-modules: starts prosody on all active nodes
MT#11123 Fixed wheezy based NGCP cannot be installed due to new MySQL 5.5.41-0+wheezy1 (error: No database selected)
MT#11069 Fixed prosody, prevent istening in all interfaces
MT#11063 [Carrier] Move PXE boot configuration to ngcp templates framework
MT#11005 cloud_pbx_callqueue on huntgroup subscriber
MT#10965 Unable to remove CFU with timeset via REST API
MT#10931 Do not generate “oldmediaip” from SDP for calls through the system
MT#10885 Prevent cascading of group deletion in ngcp-schema
MT#10879 Primary number can not be set to null if it has been already set
MT#10855 wrong monitrc configuration for iaxmodem in mr3.6.1
MT#10705 CloudPBX: Allow to set group members from within group
MT#10587 Implement basic Yealink auto-provisioning
MT#10581 motd not in etckeeper gitignore file
MT#10367 Make language for voicemail/conference sounds a user preference
MT#10337 Add update/override to cloudpbx-devices tool
MT#10315 Setting lock level on subscriber creation via REST API does not work as expected
MT#10277 Release specific deployment.sh
MT#10199 Add NCOS Level in Subscriber Profile
MT#10159 vmnotify it is not triggered if you read/delete VM from NGCP Panel
MT#9453 Some improvements order to disable call forward
MT#8591 With sst_enable=no, Session-Expires, Min-SE are not propagated but “Supported: timer” yes.
MT#8307 ngcp-fraud-auto-lock does not filter out peering contracts
MT#8129 PIN code for entrance to conference room
MT#5585 NGCP-Panel: missed column “charges” for CDRs list

Sipwise exhibits at BBWF 2019 in Amsterdam.

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Broadband World Forum 2019 (BBWF) is one of the leading events for fixed and wireless broadband ecosystems. More than 4,100 professionals in this field will gather on the 15th to 17th of October 2019 in Amsterdam. Sipwise takes part as an exhibitor and showcases its Unified Communication solutions that are tailored for telcos and Internet Service Providers (ISPs).

The focus of this year’s BBWF is all about 5G, innovations at the edge and next-generation networks.

Oliver Schlögl, Head of Sales at Sipwise summarizes: “Broadband World Forum is a wide-ranging platform to connect with global players in the field of fixed and wireless lines. To face the upcoming challenges for broadband suppliers, such as 5G, Sipwise is keen to showcase the solutions at the exhibition for interested corporations.”

“Our participation at BBWF 2019 is dedicated to the further development of business communication in the focus of comprehensive digitization and virtualization. We are excited to get in contact with visitors at our booth number A36 to discuss the significance of investing in Sipwise’s scalable technology and how it can impact corporations in terms of business expansion and profit benefits,” stated Oliver Schlögl.

For more information about the participation of Sipwise at Broadband World Forum, please check our profile at  BBWF2019.

Sipwise will be at booth A36 at RAI in Amsterdam/Netherlands. We are looking forward to welcoming you and to discuss the latest trending topics like 5G for telcos and Unified Communication solutions.

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About Sipwise

Sipwise unified communication platforms are targeted at fixed, converged and wireless service providers supporting a variety of access technologies like Cable, xDSL, FTTx, WiFi and WiMAX, as well as “Over-the-Top” operators.

With years of expertise in VoIP and UC solutions, Sipwise works with clients on all continents and is servicing over 65 Telcos in 25 countries worldwide. Sipwise offers an unbeaten price-to-performance ratio to communication service providers, with ease of integration towards their technical, OSS and BSS infrastructure, as well as outstanding flexibility.

For more information, please visit https://www.sipwise.com.

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Sipwise and ALE Holding Combine International Unified Communications Projects in Africa and Worldwide

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One year ago, the telecom group Alcatel Lucent Enterprise (ALE Holding) acquired a majority stake in the Austrian tech expert company Sipwise. Since then, the two tech companies have been working together to implement innovative Unified Communications projects in Africa and worldwide – including but not limited to the Cloud Edition phones based on the Sipwise CPBX. To reinforce their position in Africa, Sipwise offers custom-made solutions for the evolving connectivity.

08th November 2019, Brunn am Gebirge/Kornwestheim – Through the takeover by ALE Holding, Sipwise has a strong partner and can act with increased flexibility. In recent years, Sipwise has strengthened its leading position in the fixed-line telephony sector. The telephony solutions from Sipwise, which are in ever-increasing demand, help Telcos, Internet and city network operators as well as MNOs and MVNOs to offer competitive telephony products. Through ALE Holding presence, Sipwise is able to further international projects. ALE Holding also relies on the extensive expertise of the Austrian company: In the cloud and VoIP telephony sector, Sipwise is setting new standards based on hardware and software. In 2018 Sipwise entered the African market with their solutions. By offering an attractive array of products in the field of Unified Communications, connectivity, voice and communication solutions, Sipwise is supporting the well-established service providers in South Africa.

Strong partnership for international projects

Together, Sipwise and ALE Holding have created increased cooperation, to supply the telephony market with innovative solutions for the Africa market and worldwide.

Daniel Tiefnig, CEO of Sipwise stated: “With ALE Holding as our partner, we can realize innovative projects. In the future, we will cooperate even more closely with Alcatel Lucent Enterprises. This will enable us to maintain our dynamism, develop further solutions and at the same time having the stability provided by the ALE Holding”. Through this synergy, the companies support each other with new impulses. The tech corporations take this pool of ideas as an incentive and work on numerous concepts. One of the outputs of this cooperation is the “Cloud Edition Telephony” which enhances the Unified Communications portfolio in Africa – and worldwide.

Sipwise CPBX for Cloud Edition phones

Cloud telephony marks the starting signal for the joint cooperation: Sipwise’s CPBX solution is integrated into the heart of the Alcatel Lucent Enterprises Cloud Edition phone: Through this mobile-based option of a telephone system, the CPBX (Cloud Private Branch Exchange) offers a Next Generation Network (NGN) technology. The cloud telephony enables an easy exchange through the turnkey solution. The intuitive telephony with impeccable audio quality supports the everyday business with crystal-clear voice connections – even via Bluetooth headsets. Add-ons for the Cloud Edition phones form the individual design of the telephone screens in the company’s own corporate design and the connection with other service products in the cloud. The phones are available in five variations and can be seamlessly integrated into existing systems using the CPBX technology.

Cloud solution for business telephony

The CPBX is a mobile-based variation of a telephone system. Sipwise’s telecommunications solution enables simple integration into corporate infrastructures. With extensive features, the CPBX enables smooth telephony in everyday life: waiting loops, conference telephony and call forwarding. Virtual telephone switching offers small and medium-sized companies a flexible and cost-effective telecommunications system. The technology is tailored to the needs of today’s communication, offering a smooth user experience and setting itself as a reliable product through its high available technology.

About Sipwise GmbH

Sipwise GmbH is Europe’s leading provider of innovative next-generation telecommunications solutions. The company offers Voice-over-IP telephony based on hardware and software. Sipwise solutions now handle more than 100 million call minutes per month worldwide. In addition to well-known telecom providers such as Liberty Globals UPC, Sipwise is gaining more and more large, medium-sized and smaller network operators as customers. Recently, several very renowned companies have been added as new customers on the German market alone, such as T-Mobile, Kabel Plus and Tele2.

Sipwise was founded in 2008 by telecommunications professionals Daniel Tiefnig, Andreas Granig and Atilla Ceylan. Through the majority stake acquisition, Sipwise has continued to operate as a profitable business with a strong partner and has recorded increasing growth. More information can be found at www.sipwise.com

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Sipwise Enters the African Market with business VoIP solutions

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ITEC Communications PTY (ITEC) trusts in Sipwise Class 5 Softswitch as VoIP Solutions for their business.

The African continent is currently evolving rapidly in the field of communication technology. Experts predict that the number of smartphone users in South Africa will rise by around 28 percent by 2023 (which currently forecasts a 5 million user growth)[1]. This is where Sipwise and ITEC unite their power in Unified Communications. ITEC is a well-established full infrastructure service provider in the technology arena for connectivity, voice, Unified Communications and communications solutions in South Africa. The company is part of the ITEC Group and tackles the ever-changing situation in South Africa by deploying Sipwise’s proven C5 Softswitch.

Connecting experts: ITEC and Sipwise

The first meeting between ITEC and Sipwise was at the well-known exhibition in South Africa: AfricaCom. ITEC was in search of a new voice core. The company found a competent partner in Sipwise who offers a great selection of products: from Class 5 Softswitch (C5) and Cloud PBX to Sip:phone. ITEC chose Sipwise’s C5 as their leading system VoIP solution for their business. The product is a highly scalable SIP softswitch for business voice solutions. It is easily expandable by simply enabling various software modules. With a foresight for the African market, the softswitch can be deployed on 3G, 4G and – on the newest innovation – 5G mobile phones.

Connecting people: Class 5 Softswitch for ITEC

ITEC intended to implement an off the shelf system which can cover most of their requirements. Sipwise stood out from its competitors primarily because of the excellent service quality provided by the team.

Proving the company’s flexibility, the employees of Sipwise were able to assemble a custom-made package for ITEC to meet all their wants and needs. Due to an effective and united team, and based on the agreement with Sipwise, ITEC was able to get to know the softswitch in detail, gained experience with the Class 5 product in their business context and received comprehensive training from Sipwise. With the CTP (Certified Training Package), users of the CE (Community Edition) get a qualified and certified remote or on-site training.

ITEC’s intention was aimed at a switch for their voice calls, rate all CDR’s for billing, cater for number portability and provision customer voice. In addition to these, the South African company wanted to have a partner on hand who can troubleshoot and resolve potential issues with competence should any arise. Since Sipwise’s C5 Softswitch combines all the features they were looking for (and more) as well as the support they wanted, the corporation decided to ally with the Austrian company.

ITEC’s General Manager appreciates the human component above all: “Sipwise offers extensive troubleshooting possibilities. The product C5 itself is user-friendly and provides a powerful interface for a smaller operator and is stable with high availability. What stands out most is that it is about people and not just the product. Sipwise presents itself as a flexible company, where their pleasant employees are always willing to compromise and help whenever it is needed. With 24/7 support, challenges are no longer an obstacle.”

The round-the-clock support offered by Sipwise is available in English, and business interactions can be conducted in different languages, such as English, German, French, Portuguese, Spanish, Russian and more. The aim is always customer satisfaction, a highly efficient workflow and flexibility when it comes to requirements and product integrations.

Connecting with ease: C5’s integration

The outstanding features of the Sipwise C5 led to the decision of purchasing the softswitch system. The General Manager of ITEC is particularly pleased with the easy integration of the product: “We had to connect and integrate the Class 5 with our other existing systems, but with the provided API it was simple to carry out the embedding of the VoIP solution for our business purpose”.

Due to its easiness, the Sipwise C5 Softswitch is suitable for various business sizes – ranging from smaller companies to larger businesses. With ITEC’s continuing growth in their customer base, the company has a strategic goal with Sipwise in mind: to scale up to the next tier of Sipwise’s platform.

Sipwise Class 5 Softswitch is a turn-key, carrier-grade VoIP Softswitch and is the core system of our Unified Communication solutions. For further information about the product, head over to our product page or contact our sales team at sales@sipwise.com.

About Sipwise

Sipwise unified communication platforms are targeted at fixed converged and wireless service providers supporting a variety of access technologies like Cable, xDSL, FTTx, WiFi and WiMAX, as well as “Over-the-Top” operators.

With years of expertise in VoIP and UC solutions, Sipwise works with clients on all continents and is servicing over 65 Telcos in 25 countries worldwide. Sipwise offers an unbeaten price-to-performance ratio to communication service providers, with ease of integration towards their technical, OSS and BSS infrastructure, as well as outstanding flexibility.

For more information, please visit https://www.sipwise.com.

[1] https://www.statista.com/statistics/488376/forecast-of-smartphone-users-in-south-africa/

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Sipwise at Fiberdays 2020 in Wiesbaden/Germany

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Fiberdays 2020 is Germany’s leading trade fair for fibre optics and will take place on March 5 and 6. Up to 3,000 trade visitors from Germany and abroad are expected to attend the 2-day event. The fair is dedicated to one of the most relevant topics in the industry. In the past year, the trade fair has been a clear success. Sipwise exhibits and presents its solutions around cloud-based communication tailored to the glass fibre industry at the Fiberdays 2020.

Thanks to the perfect combination of exhibition, seminars, workshops and networking night, the Fiberdays offer the perfect stage to present the products and solutions of Sipwise. The event enables to showcase how a powerful infrastructure and cloud-based products can help boost growth within a company.

Oliver Schlögl, Head of Marketing and Sales at Sipwise: “Fiberdays 2019 have been overwhelming. The fair gives us the occasion to maintain some valuable business contacts in Germany and to build new business relations. The event is the right platform to present our Unified Communication Solutions in relation to fibre glass technology. Interested visitors have the unique opportunity to test our products on smartphones and tablets in real life.”

UniCom Solutions

The core of Sipwise’s solutions is Class 5 Softswitch. As a turn-key and carrier-grade VoIP-product, it is the indispensable system for Unified Communications. Internet and telephony service providers are able to offer excellent real-time communication with cloud services and VoIP. Supplemented with further products of Sipwise such as Cloud PBX or Web:RTC, Sipwise provides companies with an aligned solution of various user needs.

Sipwise will be at booth 815 at Wiesbaden in Germany. We are looking forward to welcoming you to our stand and get in touch with you. For more information visit brekoverband.de/fiberdays20-the-fiber-fair

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About Sipwise

Sipwise unified communication platforms are targeted at fixed, converged and wireless service providers supporting a variety of access technologies like Cable, xDSL, FTTx, WiFi and WiMAX, as well as “Over-the-Top” operators.

With years of expertise in VoIP and UC solutions, Sipwise works with clients on all continents and is servicing over 65 Telcos in 25 countries worldwide. Sipwise offers an unbeaten price-to-performance ratio to communication service providers, with ease of integration towards their technical, OSS and BSS infrastructure, as well as outstanding flexibility.

For more information, please visit https://www.sipwise.com.

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Sipwise at ASLAN 2020 in Madrid/Spain

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ASLAN 2020 is the leading exhibition in regards to digital transformation in Spain. The fair gathers professionals of the IT-industry from 10th to 11th of March 2020 in Madrid at IFEMA. Due to last year’s overwhelming success, Sipwise will participate again this year.

Sipwise will welcome you at booth 87. With a broad knowledge about Unified Communications, the team is eager to discuss the latest news and solutions with you.

Product Launch: Certified Training Package for CE

Sipwise’s new product innovation is being presented at #ASLAN 2020 – Certified Training Package (CTP):

Users of the Community Edition (CE) may attend an online course from certified instructors. With CTP, users are allowed to tap the full potential of the Open-Source Software for CE to be prepared for daily use and possible challenges.

Still, the core of Sipwise’s solutions at ASLAN 2020 is the Class 5 Softswitch. As a turn-key and carrier-grade VoIP-product, it is the indispensable system for Unified Communications. Internet and telecommunication service providers will be able to offer excellent real-time communication with cloud services and VoIP. Accompanied by complementary products of Sipwise such as Cloud PBX or Sip:Phone app, Sipwise offers companies an aligned solution for various user needs.

The exhibition offers a wide stage to showcase products and solutions from Sipwise to interested service providers who are operating in the telecommunications sector. 

For more information and complimentary tickets, please click on this link.


About Sipwise

Sipwise unified communication platforms are targeted at fixed, converged and wireless service providers supporting a variety of access technologies like Cable, xDSL, FTTx, WiFi and WiMAX, as well as “Over-the-Top” operators.

With years of expertise in VoIP and UC solutions, Sipwise works with clients on all continents and is servicing over 65 Telcos in 25 countries worldwide. Sipwise offers an unbeaten price-to-performance ratio to communication service providers, with ease of integration towards their technical, OSS and BSS infrastructure, as well as outstanding flexibility.

For more information, please visit https://www.sipwise.com.

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Sipwise CEO statement on the current COVID-19 crisis

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Dear customers, partners and stakeholders,

In response to the rapid global spread of COVID-19, governments around the world have implemented action plans to enable containment. Proportionally to the spread of the virus, the measures taken by governments worldwide are intensifying.

We understand that these developments may cause concern and would therefore like to inform you about the current situation at Sipwise.

Precautionary measures at Sipwise due to COVID-19

To keep our employees, customers, partners and their families healthy and safe, we have been implementing precautionary measures in the last weeks, specifically regarding our office and client-relations-policy.

Currently, no business trips are carried out, neither nationally nor internationally. Our trade fair appearances and visits, even at events that have not yet been postponed or cancelled, are suspended until further notice.

To ensure the safety of our employees, we have completely switched to remote working. Since Monday 16th of March employees are only present in the head office in exceptional cases. All internal and external meetings have been converted to technology-supported online conferences carried out via Rainbow – the collaboration platform of our parent company Alcatel-Lucent. Click here to read more.

We are in the gratifying situation that no COVID-19 case or suspected case has been reported within our workforce to date.

Our service continues

Our unified communications solutions are not affected by the internal measures taken. Our servers are fully functional, our support team continues to work for you, developers and product managers continue to work on incremental improvements to our product portfolio and our management team are 100% committed to the performance of our people and our technology. You can be sure that there will be no limitations in the availability and continuity of our solutions.

Sipwise is serving infrastructure in 31 countries on 6 continents for more than 100 commercial deployments at telco operators. We believe that communication plays a central role in overcoming this crisis, therefore we are absolutely committed to contributing our share in coping with this unprecedented situation.

As a wholly-owned subsidiary of Alcatel-Lucent Enterprise, Sipwise ensures unrestricted productivity with the company’s own Unified Communications and collaboration platform Rainbow.  In response to the COVID-19 crisis, Alcatel-Lucent is providing three months of free access to the software’s business license. More information can be found below.

Sipwise is taking every precaution to keep employees, partners and their families safe and is going out of its way to keep it’s clients fully operating until this pandemic has passed.

We encourage everyone to follow precautionary measures communicated by public health authorities.

At the same time, we will continue to support and service you to the best of our ability.

Sincerely,

Daniel Tiefnig, CEO Sipwise

Our offering | 3 months free Enterprise licenses for Alcatel-Lucent-Rainbow!

Not only our clients but also every company that needs their employees to work remotely requires a fully mobile, effective, ubiquitous and easy-to-use technology that ensures the integrity and security of everyone’s data or personal information.

To accommodate these requirements, Alcatel-Lucent Enterprise is offering free Rainbow Enterprise licenses for three-months (from 9th of March to the 30th of June 2020) so that businesses can experience its cloud-based communication platform and enable their staff to productively and securely work remotely.

With the three-month trial period commencing on the date of activation, businesses will be able to immediately use Rainbow on the Web, their mobile, desktop and tablet.  With no additional hardware or software installations required, in one click employees can get started immediately.

At the end of the three-month promotional period, one will be able to extend one’s subscription and continue to benefit from the tried-and-test communication and collaboration platform.

To benefit from our 3 months’ free Enterprise licenses*, click here.

About Sipwise

Sipwise unified communication platforms are targeted at fixed, converged and wireless service providers supporting a variety of access technologies like Cable, xDSL, FTTx, WiFi and WiMAX, as well as “Over-the-Top” operators.

With years of expertise in VoIP and UC solutions, Sipwise works with clients on all continents and is servicing over 100 commercial deployments and more than 60 Telcos. Sipwise offers an unbeaten price-to-performance ratio to communication service providers with ease of integration towards their technical OSS and BSS infrastructure, as well as outstanding flexibility.

For more information, please visit https://www.sipwise.com.

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Great news for Sipwise CE Users: we are releasing Certified Training Package (CTP)

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On a daily basis, we listen to our community with care and curiosity. Many of our Sipwise Community Edition (CE) users have been struggling to unleash the full potential of the Sipwise CE and the impact it can have on their business. We drew our conclusions, reacted according to the feedbacks and processed it – Your story allows us to innovate and therefore, we created the Certified Training Package (CTP). We want our users to reap the benefits of official Sipwise training.

The CTP is a two-day online course that Sipwise has made available to Sipwise CE users. During the training, the participants will be guided on how to better use our Sipwise C5 CE to its full performance level. With CTP, the CE users will be able to tap the maximum potential of the open-source software and will be prepared for daily challenges.

How can Community Edition’s users benefit from the Certified Training Package (CTP)?

All participants will be instructed on how to handle the Community Edition (CE) with efficiency to conquer upcoming troubleshooting scenarios in the future with ease and to find solutions adapted to their business cases. The CTP will be held by a certified and active Sipwise engineer who will provide up-to-date information on our solutions and procedures. In the Q&A session, the attendees will have the opportunity to discuss individual topics and get professional feedback.

The first-day session will cover topics such as architecture, basic and advanced configuration; call flows, backup and troubleshooting-issues. The second-day session consists of a series of hands-on activities will be conducted to consolidate the knowledge and to assure the day-to-day operations with the Sipwise CE will be optimized.

CTP for Sipwise Training: Key facts and figures

Do you still miss information? Here are the key facts in a nutshell:

  • Languages: CTP is available in the following languages: English, Italian, Russian and Spanish
  • Participants: minimum of 2 participants and up to 14 Participants per session.
  • Location:
    • online
    • Sipwise offers onsite-training upon request (additional fees apply)
  • Trainer: a certified and active Sipwise engineer will hold the sessions. Together with the trainer, the participants will dive deep into the possibilities the Community Editions (CE) offers and gain experience with our open-source platform.

Do you want specifics? Please visit https://sipwise.com/products/sipwise-ctp-certified-training-package.

About Sipwise

Sipwise’s unified communication platforms are targeted at fixed, converged and wireless service providers supporting a variety of access technologies like Cable, xDSL, FTTx, WiFi and WiMAX, as well as “Over-the-Top” operators.

With years of expertise in VoIP and UC solutions, Sipwise works with clients on all continents and is servicing over 100 commercial deployments and more than 60 Telcos. Sipwise offers an unbeaten price-to-performance ratio to communication service providers with ease of integration towards their technical OSS and BSS infrastructure, as well as outstanding flexibility.

For more information, please visit https://www.sipwise.com.

The post Great news for Sipwise CE Users: we are releasing Certified Training Package (CTP) appeared first on Sipwise.

Sipwise releases Media Gateway (MG) developed in close cooperation with Sangoma.

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As a result of increased demand, Sipwise is launching the Sipwise Media Gateway, a service that has been developed in close cooperation with Sangoma. The Sipwise MG is connecting IP-Networks to Public Switched Networks via high-density optical telephony and Ethernet connections in a compact 1U form.

Media Gateway: A hybrid solution before taking the final leap to Next-Generation Communication systems

Do you not want to change your existing ISDN infrastructure just yet, but still want to take advantage of VoIP telephony benefits?

Stability, cost savings, simultaneous use of any telephone numbers and terminals, independence of location, as well as central communication via one network, the Internet, instead of operating a telephone network in parallel – these are some of the many advantages of VoIP/IP telephony.

The Media Gateway is an extension of the Sipwise Unified Communications portfolio and supports telecommunications providers switching from TDM/SS7 to SIP/IP.

What are Media Gateways? An executive summary.

Media Gateways are a network transition used in telecommunications, next-generation networks and IP telephony. They serve as links that enable communication between different and not always compatible systems. These systems can be networks, single computers or different applications on computers.

The Sipwise MG converts analogue media and signalling to support efficient and reliable voice, fax and multimedia sessions for mobile and cloud-based applications. It secures sessions across IP and mixed network boundaries to support service level agreements.

What are the capabilities of the Sipwise Media Gateway?

  • Deployment and outstanding reliability
  • Scripting and Modification Suitability (employs Sangoma’s Programmable Protocol Language)
  • Multiple algorithms for routing and disposition
  • Pre Call-Announcement for branding employment
  • Hardware-less transcoding via integrated encryption and transcoding support
  • Scalable up to 2250 SIP sessions and 2016 channels of SS7 signalling

Do you want more information? Please visit https://sipwise.com/products/ sipwise-mg-media-gateway-for-service-providers.

About Sipwise

Sipwise’s unified communication platforms are targeted at fixed converged and wireless service providers supporting a variety of access technologies like Cable, xDSL, FTTx, WiFi and WiMAX, as well as “Over-the-Top” operators.

With years of expertise in VoIP and UC solutions, Sipwise works with clients on all continents and is servicing over 100 commercial deployments and more than 60 Telcos. Sipwise offers an unbeaten price-to-performance ratio to communication service providers with ease of integration towards their technical OSS and BSS infrastructure, as well as outstanding flexibility.

For more information, please visit https://www.sipwise.com.

The post Sipwise releases Media Gateway (MG) developed in close cooperation with Sangoma. appeared first on Sipwise.

Sipwise releases ALE Visual Auto-Attendant (VAA) as flexible call flow engine for easy routing and call management.

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For companies – ranging from smaller to larger businesses – a telephone call is the first point of contact. Sipwise now released a new product for easy routing calls and call management within an enterprise: The Alcatel Lucent Enterprise (ALE) Visual Auto-Attendant (VAA). It is a graphically designable call flow engine for the Sipwise C5 Softswitch (NGCP). The extending feature is a powerful business routing and calls answering/IVR (Interactive Voice Response) module.

VAA as an Easy Call Management and personalization tool

Ever called a company and been put on hold in the waiting loop? No matter with which request a customer or business partner contacts an enterprise, the caller never wants to hear one thing: a busy signal.

The VAA from Sipwise provides professional handling of calls with a virtual receptionist that enables a 24/7 availability without the intervention of a staff member. This way, the caller gets around-the-clock assistance – no matter of the time the person dials the company’s number. The telephone call starts off with a kind greeting and opens the conversation with the company’s jingle or announcements from a pleasant voice. The caller is immediately drawn into the company’s brand world. Sipwise’s VAA offers the opportunity of a personalized greeting by integrating the database and contact centre of an enterprise into the VAA.

For even more convenience, the software allows an IVR option with SQL and HTTP connector which routes the calls based on information of the company’s database. The VAA simplifies the process of a telephone call for both sides – for the caller and the company: The easy routing directs the caller automatically to the desired employee, department or its voicemail with TTS (Text to Speech) and ASR (Automatic Speech Recognition). The VAA recognises the typed or spoken menu system (e.g. “press/say 1 for service etc.) and routes the caller to mobile phones, landline phones or VoIP phones numbers.

VAA call management with a user-friendly interface

The mechanism of the VAA is an easy to use software. The graphical interface enables simple management by drag-and-drop building blocks to create call routing based on business hours and (holiday) calendars. As an intuitive software, the VAA does not require any high-level IT-skills and therefore, reduces the training time for the personnel.

The web interface – which includes a dashboard and reporting area – can be used by multiple customers. Each of them has its own private view which enables resellers multi-tenant features for their customers. With the script block, companies can add additional functionalities based on their business needs.

The VAA software is scalable as it is ready to use for smaller and larger multi-site organizations and offers SIP-based connectivity to communication servers.

About Sipwise

Sipwise unified communication platforms are targeted at fixed, converged and wireless service providers supporting a variety of access technologies like Cable, xDSL, FTTx, WiFi and WiMAX, as well as “Over-the-Top” operators.

With years of expertise in VoIP and UC solutions, Sipwise works with clients on all continents and is servicing over 100 commercial deployments and more than 60 Telcos. Sipwise offers an unbeaten price-to-performance ratio to communication service providers with ease of integration towards their technical OSS and BSS infrastructure, as well as outstanding flexibility.

For more information, please visit https://www.sipwise.com.

The post Sipwise releases ALE Visual Auto-Attendant (VAA) as flexible call flow engine for easy routing and call management. appeared first on Sipwise.

Sipwise C5 is easy to migrate and scale-up

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“Everybody working at Sipwise operations is very excited about the ease and fluency of migrating platforms. With every successful service delivery, we only admire our flagship product more: Sipwise C5 — carrier-grade VoIP Softswitch“, the Sipwise Operations team states.

“Not so long ago, we migrated production VoIP services from Sipwise C5 Pro to C5 Carrier (more powerful and scaleable version) platforms”, says Head of Operations Daniel Grotti. Below, the Sipwise Ops team explains how the migration was executed.

How did the Sipwise C5 Pro to C5 Carrier migration happen?

Both platforms have the same structure: the same OS and processes running, the similar configuration of components, however the architecture is different:

The Sipwise C5 Pro has only two physical servers running in Active-StandBy mode (both servers have the same services, but they run only on the active server).

The 5 Steps for migration of production services:

  1. IP addresses swap preparation
  2. Database copy to the new platform
  3. IP addresses swap
  4. Basic functionality tests
  5. Copy of additional data from the old platform (CDR delta as the old platform had subscriber calls while the DB was copied already, etc.)

Most of the time, the teams waited until the huge database gets copied. Also, the customer performed comprehensive tests within an hour to ensure that the service worked fine on the new platform and no roll-back (that would have taken a few minutes to swap the IP addresses back) was necessary.

Similarly, the Sipwise Professional Services department migrates data and services from old hardware to the new hardware and even from old releases to the new ones (with database schema and data update).

“We love how the Sipwise C5 can be deployed, configured, maintained and even migrated in a secure, straightforward and reliable way”, concludes the Sipwise Operations Team.

About Sipwise

Sipwise’s unified communication platforms are targeted at fixed, converged and wireless service providers supporting a variety of access technologies like mobile radio, Cable, xDSL, FTTx, WiFi and WiMAX.

With years of expertise in VoIP and UC solutions, Sipwise works with clients all over the world and is supporting over 100 commercial deployments by MNOs, MVNOs, Telcos, cable network providers, utilities and “Over-the-Top”- operators.
For more information, please visit https://www.sipwise.com.

The post Sipwise C5 is easy to migrate and scale-up appeared first on Sipwise.

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