Voice over Internet Protocol (VoIP) / Asterisk / SIP

Most significant commercial experience was on Regus' CallStream development team. When the technical lead left I was also tasked with product development and (global) 3rd line support for 3000+ asterisk installs and 300000 telephones and interfacing with the Regus global networks team.

Product development included working with Clavister on an ALG, SIP in a NAT configuration and connecting other products to the Asterisk-based PBX platform running a combination of SIP and PRI for backhaul.

Over the last few years I've consulted to a number of local and multinational Companies on a once off basis regarding Asterisk Issues:

  • a French/UK company with a native install experiencing call dropouts that I diagnosed to be echo cancellation running on both SIP handset and Digium PRI cards interfering with each other.
  • Migrating a collection of small UK companies from a SIP Trunk -> POTS (in the office set up) to a full SIP setup using 2 hosted Asterisk PBXs installed from source with multiple SIP providers and Trunks including integrating a LAMP-based backoffice application with their new Digium D50 handsets. I have recently done the prototype work on an Android App that would integrate with this solution too.
  • A Portuguese/UK/multinational with number presentation issues using FreeSWITCH including advising on .NET code changes necessary with interfacing software.

Asterisk specific skills

  • Realtime config, using direct database access and accessing heavily processed external data via curl. 
  • Dial from Application, CLI based click to dial from PHP to support any SIP end point and specifically using the SNOM API for 300 and 320 handsets.
  • Caller ID Lookup using Asterisk curl to obtain lookup information from a PHP webservice (MySQL, MSSQL and XML file)
  • Source compiled, IVR (greetings and handling out of hours), Call Recording, Voice Mail (delivered via email attachment), multi-party conference bridge, Ring Groups, DDI / DDO

 ALG / SIP Proxy / NAT

OpenSIPS on Linux to protect an internet facing Asterisk install providing an end point for URI-based call routing and detecting and feeding sources of abusive traffic dynamically to a firewall.

Claviser ALG on firewalls used routers

Using Asterisk as a SIP gateway to ININ's I3 platform

Trunks / VoIP / SIP

  • Gradwell - IAX & SIP
  • Vodafone
  • Colt
  • L3
  • Orbtalk, Solar(Gamma) & Sipgate - SIP
  • BT - ISDN2e
  • Inter-Asterisk comms trunking UK and ZA Asterisk IP PBXs together
  • Call quality rating, using RTP data - most specifically, but not limited to, the G.729 codec
  • Reatime call cost rating using 33k cost routes from Telco with appropriate number (re)formatting / matching and caching of route costs.

Android as a SIP client

I've worked on some investigation using the Galaxy S3 as a serious SIP client.

During the time at World First (until 2011) I was responsible for negotiations with Fixed Line and SIP trunk providers and ultimately responsible (albeit at arms length) for the entire multi-national (Nortel, then Shoretel) telephone system (UK, AU and DE).

My introduction to IP Telephony was during a 2008/9 contact position in Soho, looking after VoIP sites from 5 to 50 SIP extentions mostly in production houses.  I was soley responsible for the installation and setup of Asterisk and for provisioning Snom phones using DHCP and an external provisioning server.

Happy to work as an Employee, Consultant, Contractor / Self Employed or via Limited Company