You need to gather the data and file it on JIRA http://jira.freeswitch.org/ Include all details including a log with full sip trace enabled. Do these commands in fs_cli sofia global siptrace on sofia loglevel debug (/log_debug in fs_cli) reproduce your problem and attach the trace. On Wed, Feb 15, 2012 at 8:37 AM, Daniel Knaggs < Daniel.Knaggs@realitysolutions.co.uk> wrote:
Hello,****
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We’ve recently switched from Asterisk to FreeSwitch and experiencing a minor problem.****
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Currently we’ve got twenty LinkSys/Cisco phones (12x SPA942, 6x SPA962, and 2x SPA504) and experiencing dropped calls when transferring calls.****
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This seems to happen more often when the caller has hit the xfer button for the second time before the callee has even answered the phone. But sometimes drops no more than one call a day if the caller waits until the callee has answered.****
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**· **All phones are running the latest firmware (v6) and have been factory reset****
**· **FreeSwitch is running on version “1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)”****
**· **RTP packet size has been changed to 0.020 as per wiki for all phones****
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Has anyone got any ideas?****
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Thanks in advance.****
Daniel Knaggs
Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs@realitysolutions.co.uk www.realitysolutions.co.uk ------------------------------ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire
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