Voice over Internet Protocol (VoIP) / Asterisk / SIP
From January 2014 the Voice project I was working on at Regus lost it's technical lead and I was increasingly asked to pick up networks and telephony planning and the project's product support. This involved significant knowledge of SIP, external connections via an ALG, SIP in pure NAT and connecting other products via the Asterisk-based PBX platform distributed through around 1500 sites globally running a combination of SIP and PRI.
Over the last few years I've consulted to a number of local and multinational Companies on a once off basis regarding Asterisk Issues. Notably:
- 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.
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
OpenSIPS on Linux
Claviser ALG routers
Using Asterisk as a SIP gateway to ININ's I3 platform
Trunks / VoIP / SIP
- Gradwell - IAX & SIP
- 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