[asterisk-bugs] [JIRA] Feedback Requested: (ASTERISK-20174) Asterisk becomes unresponsive when tryung to send fax with T38

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-20174?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-20174:
------------------------------------

    Assignee: Andrew Nowrot
      Status: Waiting for Feedback  (was: Triage)

> Asterisk becomes unresponsive when tryung to send fax with T38
> --------------------------------------------------------------
>
>                 Key: ASTERISK-20174
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20174
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/T.38
>    Affects Versions: 10.6.1
>         Environment: Debian wheezy witj kernel 3.1.6 no PSTN cards
>            Reporter: Andrew Nowrot
>            Assignee: Andrew Nowrot
>            Severity: Critical
>
> When trying to send fax with T38 between two fax devices connected to SIP ATA (SPA2102) my asterisk becomes unresponsive. ATA has a parameter "FAX Tone Detect Mode" it comes with three choices "caller and callee", "caller only", "callee only".  When it is set to something other than "callee only" and I try to send a fax, asterisk freezes. It stops responding to anything, it is not processing calls and CLI is completely unresponsive. With "calee only" everythig works like charm. Is this a bug?
> I am sending faxes from a regular fax device to a machine which gives me the choice between fax transmission and leaving the voice message, so my side (calling) has to initiate the fax transmission. According to https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway only callee should initiate the T38 transmission and that is the properly configured endpoint. But when caller initiates the fax transmission Asterisk should not crash.
> When doing sip debug this is the last line and after that asterisk freezes
> Got T.38 offer in SDP in dialog 341bae0d51a1519967000e3f32c49145 at xxx.xxx.xxx.xxx
> Capabilities: us - (alaw|slin), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
> Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

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