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Smartvox Professional Services:

Are you trying to install a solution yourself but got stuck and need some professional support? Technical assistance from Smartvox may be a more cost effective solution than you realise when you take into account the true cost of your time.

Our professional services division can provide consultancy, expert configuration and support of Vegastream, Asterisk, SER, OpenSER or OpenSIPS systems even if they were not sourced from Smartvox. Typical price in GBP for consultancy is £45/hour plus travel expenses. In other currencies, this equates approximately to USD $75/hour or Euros €50/hour.

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SIP, Asterisk, OpenSIPS: Frequently Asked Questions

This section is under continuous development as new solutions and examples are added. Please check regularly to see if we have added more links.

Understanding SIP:

The following are links to articles in the Smartvox Knowledgebase: Can you make free calls using Voice over IP (VoIP)? How do you choose the best VoIP Service Provider? SIP Servers explained SIP Registration explained What is the "SIP Proxy" setting on an IP phone? What is the "Outbound Proxy Server" setting on an IP phone? SIP and NAT: Why is it a problem? SIP devices behind NAT: What solutions are available? What options are there for High Availability and Failover with SIP?

Asterisk:

The articles here are based on many year's experience working with Asterisk. Mostly, the examples discussed here describe how you configure Asterisk by direct editing of the CONF files. This information may also be useful if you are using FreePBX, Trixbox, AsteriskNow or other GUI configuration utilities. The following are links to articles in the Smartvox Knowledgebase: How must Asterisk be configured when it is behind NAT? How secure is your Asterisk PBX? Part 1: Asterisk vulnerabilities and the tactics used to hack it How secure is your Asterisk PBX? Part 2: Measures you can take to protect your Asterisk PBX How secure is your Asterisk PBX? Part 3: SIP Domains, trunk-to-trunk features and DoS protection Using SIP Trunks with Asterisk - Part 1: What are SIP Trunks? Using SIP Trunks with Asterisk - Part 2: Registration, authentication and inbound connections Using SIP Trunks with Asterisk - Part 2: Outbound connections, DID/DDI matching and fault finding How do you configure Asterisk to support IP phones as extensions? Configuring a shared voicemail box and linking it to a BLF key on your IP phone Using a Custom Device State to activate a BLF lamp when messages are waiting in the shared box What is the SIP Subscribe/Notify mechanism and how do Asterisk hints relate to it? What are SIP Domains and how do you use them in Asterisk? VoIP QoS Settings - Part 1: Overview and explanation of Layer 2 QoS settings VoIP QoS Settings - Part 2: An explanation of Layer 3 QoS settings including DSCP,DiffServ,ToS VoIP QoS in practice - Explores how QoS influences the way network congestion is handled Shared Line Appearances (SLA) on Asterisk - Part 1: Introduction Shared Line Appearances (SLA) on Asterisk - Part 2: sla.conf and phone configuration Shared Line Appearances (SLA) on Asterisk - Part 3: SLA hints and dial plans Shared Line Appearances (SLA) on Asterisk - Part 4: Conclusions about Asterisk SLA How does Asterisk handle Caller ID? Part 1: Introduction - CLI, ANI, From, RPI and PAI headers How does Asterisk handle Caller ID? Part 2: How Asterisk handles PAI, RPI and Privacy headers Codec selection and RTP pass-through mode in Asterisk How can I improve the resilience of my Asterisk service or introduce failover?

OpenSIPS:

OpenSIPS is a very popular, powerful and flexible Open Source SIP Proxy solution. The following are links to articles in the Smartvox Knowledgebase: An overview of OpenSIPS history and functionality - what it is and what it does What is the difference between OpenSIPS and Asterisk - when should you use each product Installing OpenSIPS v1.6.1 on CentOS 5 64 bit Installing Mediaproxy 2 on CentOS 5 64 bit Clustering OpenSIPS for High Availability - Part 1: Sharing common resources between two servers Clustering OpenSIPS for High Availability - Part 2: Far-end NAT traversal using two servers Clustering OpenSIPS for High Availability - Part 3: Using a Virtual IP for Active/Standby switching

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