diff options
| -rw-r--r-- | channels/chan_sip.c | 31 |
1 files changed, 18 insertions, 13 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index fa70dc807..793e14179 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -26331,6 +26331,24 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint if (!ast_strlen_zero(referred_by)) { pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", referred_by); } + + /* When a call is transferred to voicemail from a Digium phone, there may be + * a Diversion header present in the REFER with an appropriate reason parameter + * set. We need to update the redirecting information appropriately. + */ + ast_channel_lock(p->owner); + sip_pvt_lock(p); + ast_party_redirecting_init(&redirecting); + memset(&update_redirecting, 0, sizeof(update_redirecting)); + change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE); + + /* Do not hold the pvt lock during a call that causes an indicate or an async_goto. + * Those functions lock channels which will invalidate locking order if the pvt lock + * is held.*/ + sip_pvt_unlock(p); + ast_channel_unlock(p->owner); + ast_channel_update_redirecting(current.chan2, &redirecting, &update_redirecting); + ast_party_redirecting_free(&redirecting); } sip_pvt_lock(p); @@ -26378,20 +26396,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint } ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */ - /* When a call is transferred to voicemail from a Digium phone, there may be - * a Diversion header present in the REFER with an appropriate reason parameter - * set. We need to update the redirecting information appropriately. - */ - ast_party_redirecting_init(&redirecting); - memset(&update_redirecting, 0, sizeof(update_redirecting)); - change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE); - - /* Do not hold the pvt lock during a call that causes an indicate or an async_goto. - * Those functions lock channels which will invalidate locking order if the pvt lock - * is held.*/ sip_pvt_unlock(p); - ast_channel_update_redirecting(current.chan2, &redirecting, &update_redirecting); - ast_party_redirecting_free(&redirecting); /* For blind transfers, move the call to the new extensions. For attended transfers on multiple * servers - generate an INVITE with Replaces. Either way, let the dial plan decided |