[Asterisk-Dev] SIP transfer using the REFER method according to RFC3515

[Lea]

The function sip_transfer (see below) apparently uses the 
transmit_refer function to transfer SIP calls after they have 
been answered, using the REFER method according to RFC-3515.
However no REFER requests are evident in SIP debug output when 
Asterisk transfers calls.  Why?

Maybe this function is not used at all or depends on come conf 
setting...

< sip_transfer: Transfer SIP call (Definition at line 02605 of 
file chan_sip.c.) >
[code]02605 static int sip_transfer  (  struct ast_channel *  
ast, const char *  dest )  [static] 
02606 {
02607    struct sip_pvt *p = ast->tech_pvt;
02608    int res;
02609 
02610    ast_mutex_lock(&p->lock);
02611    if (ast->_state == AST_STATE_RING)
02612       res = sip_sipredirect(p, dest);
02613    else
02614       res = transmit_refer(p, dest);
02615    ast_mutex_unlock(&p->lock);
02616    return res;
02617 }[/code]

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