Hi all,
I'm afraid I bug in FreeSWITCH SIP call handling.
The scenario is as follows:
Very simple dialplan:
<extension name="E1">
<condition field="destination_number" expression="^(23)$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="sofia/default/$1%$${domain}"/>
<action application="respond" data="181 Call is being forwarded"/>
<action application="export" data="[EMAIL PROTECTED]
{domain};reason=unavailable"/>
<action application="transfer" data="22"/>
</condition>
</extension>
<extension name="E2">
<condition field="destination_number" expression="^(22)$">
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="sofia/default/$1%$${domain}"/>
</condition>
</extension>
User status:
Users 21 and 22 are registered, user 22 not.
Call flow:
User 21 calls number 23, recieves back 181 and phone 22 starts ringing
(Diversion header is properly appended).
Than user 22 answers (sends 200 OK and recives ACK) and after short
conversation hangs up.
And user 21's call leg isn't hanged up by Freeswitch (BYE is not sent).
After few seconds user 21 hangs up manualy and recieves 200 OK instead
of 481 Call Leg/Transaction Does Not Exist...
Is it a bug or am I doing something wrong?
Best regards,
kokoska.rokoska
PS: If someone interested I have pcap dump of communication.
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