We need assistance from an experienced VoIP professional to configure call forwarding settings for our SIP trunks. Our business utilizes an Asterisk-based phone system connected to multiple SIP providers for outgoing and incoming calling. Recently we have experienced issues with certain calls not forwarding properly to mobile or remote employees as configured.
An audit of our dial plan and call routing scripts is required to diagnose any flaws in how calls are being handled and directed. The successful candidate will have extensive familiarity with Asterisk and deep understanding of how call routing policies and SIP signaling interact. Temporary testing SIP accounts may need to be utilized to simulate various call scenarios and validate configurations are functioning as intended.
Once areas of conflict or error are identified, modifications to our dial plan syntax, call handling scripts or related configuration files must be implemented. Thorough documentation of all analyses, findings and solution implementation is expected. Comprehensive testing will then be performed across our SIP trunks and phones to ensure call forwarding to alternate contacts functions seamlessly in all intended situations.
Preference will be given to candidates holding the ATA Certification and with proven experience remediating complex VoIP setups involving multiple SIP providers and technologies like call queuing, logging and interactive voice response systems. Strong troubleshooting skills and methodical work approach is essential.
