# Asterisk 13 & MixMonitor **URL:** \ **Category:** Asterisk Support\ **Created:** [December 2, 2015, 10:43am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036 "2015-12-02T10:43:36Z")\ **Posts on this page:** 10\ **Page:** 1
**Author:** ![barmoleg](https://avatars.discourse-cdn.com/v4/letter/b/e19adc/32.png) [@barmoleg](https://community.asterisk.org/u/barmoleg)\ **Post date:** [December 2, 2015, 10:43am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/1 "2015-12-02T10:43:36Z")
Hello! MixMonitor stop recording when attended transfer call! AUDIOHOOK\_INHERIT - is deprecated! Please tell me what i can do? ---
**Author:** ![ambiorixg12](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ambiorixg12/32/1887_2.png) [@ambiorixg12](https://community.asterisk.org/u/ambiorixg12)\ **Post date:** [December 3, 2015, 3:29am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/2 "2015-12-03T03:29:10Z")
As you already said AUDIOHOOK\_INHERIT - is deprecate, but according to the Asterisk Wiki > [@](#): > > “Due to architectural changes in Asterisk 12, audiohook inheritance is performed automatically and this function now lacks function”. [wiki.asterisk.org/wiki/display/ … OK\_INHERIT](https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_AUDIOHOOK_INHERIT) ---
**Author:** ![barmoleg](https://avatars.discourse-cdn.com/v4/letter/b/e19adc/32.png) [@barmoleg](https://community.asterisk.org/u/barmoleg)\ **Post date:** [December 3, 2015, 4:46am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/3 "2015-12-03T04:46:16Z")
I understand! But i need record calls after attended transfer! How i can do it? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [December 3, 2015, 8:01am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/4 "2015-12-03T08:01:28Z")
I’m not sure if audiohooks were ever supposed to inherit across a transfer; transfers are not inheritance cases in the normal Asterisk sense. However there are many different ways of initiating an attended transfer: hook flash on dahdi, feature code, SIP enquiry and REFER. Note that in the case of SIP transfer, Asterisk has no idea that there is a transfer in progress until it gets the REFER, so will never record the enquiry as part of the original call; it is simply a new call on the phone’s “second line”. ---
**Author:** ![barmoleg](https://avatars.discourse-cdn.com/v4/letter/b/e19adc/32.png) [@barmoleg](https://community.asterisk.org/u/barmoleg)\ **Post date:** [December 3, 2015, 8:47am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/5 "2015-12-03T08:47:56Z")
SIP client send REFER when transfer call - then record stop! ---
**Author:** ![barmoleg](https://avatars.discourse-cdn.com/v4/letter/b/e19adc/32.png) [@barmoleg](https://community.asterisk.org/u/barmoleg)\ **Post date:** [December 7, 2015, 3:30am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/6 "2015-12-07T03:30:01Z")
Help please! %( ---
**Author:** ![ambiorixg12](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ambiorixg12/32/1887_2.png) [@ambiorixg12](https://community.asterisk.org/u/ambiorixg12)\ **Post date:** [December 7, 2015, 5:33am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/7 "2015-12-07T05:33:32Z")
I made an small research and some testing with Different Asterisk version, and AUDIOHOOK INHERIT() Function seems to be the solutions to this issue on the past, But not longer working since asterisk 12.If I find a workaround to this issue I will post it [issues.asterisk.org/jira/browse/ASTERISK-18220](https://issues.asterisk.org/jira/browse/ASTERISK-18220) [viewtopic.php?p=162949](http://forums.digium.com/viewtopic.php?p=162949) [wiki.asterisk.org/wiki/display/AST/Audiohooks](https://wiki.asterisk.org/wiki/display/AST/Audiohooks) [forums.asterisk.org/viewtopic.php?p=166399](http://forums.asterisk.org/viewtopic.php?p=166399) ---
**Author:** ![barmoleg](https://avatars.discourse-cdn.com/v4/letter/b/e19adc/32.png) [@barmoleg](https://community.asterisk.org/u/barmoleg)\ **Post date:** [December 8, 2015, 3:47am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/8 "2015-12-08T03:47:23Z")
Snk! But it’s not correct! In Asterisk \> 12 function is DEPRECATED! Please help! ---
**Author:** ![ambiorixg12](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ambiorixg12/32/1887_2.png) [@ambiorixg12](https://community.asterisk.org/u/ambiorixg12)\ **Post date:** [December 8, 2015, 3:52am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/9 "2015-12-08T03:52:24Z")
A quick workaround would be merge the 2 recording files using SOX tool after the second recording was completed, using variable inheritance and the against the variable MIXMONITOR FILENAME ---
**Author:** ![barmoleg](https://avatars.discourse-cdn.com/v4/letter/b/e19adc/32.png) [@barmoleg](https://community.asterisk.org/u/barmoleg)\ **Post date:** [December 8, 2015, 8:22am UTC](https://community.asterisk.org/t/asterisk-13-mixmonitor/50036/10 "2015-12-08T08:22:26Z")
A call to B - record fine B call to C - record fine When B hangup send SIP REFER, records stop and i don’t have A and C record