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Type:
Bug
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Status: Closed
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Priority:
Minor
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Resolution: Not an issue
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Affects Version/s: 14
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Fix Version/s: None
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Component/s: Core
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Labels:None
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Bug Tracker:Customer Issue
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ToDo:
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Asterisk Version:13.29.2
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Distro Version:14.0.13.24
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Distro:FreePBX Distro
Core 14.0.28.35
FreePBX Framework 14.0.13.24
When anyone attempts to transfer a call to another extension, it goes directly to voicemail and we see the following in the log:
WARNING[16707][C-0000a98a] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '<token>', expecting $end; Input:
WARNING[16707][C-0000a98a] ast_expr2.fl: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
The environment is a mixture of SIP and PJSIP devices. Is there an incompatibility between those channel drivers?