Realtime Mysql Mix Monitor not recording my calls

i have edited my extension table in mySQL to record all calls by mix monitor on a particular extension (in my case 03111308308). when this extension is called, the call should get recorded. but it is not working. I am sharing mySQL statement which i used to insert into extensions table.

insert into extensions(id,context,exten,priority,app,appdata) values (9,‘my-sip’,‘03111308308’,1,‘Dial’,‘SIP/${EXTEN},30,Ttr’),(10,‘my-sip’,‘03111308308’,2,‘MixMonitor’,’${recording}.wav, b’);

Please help.

The priority dictates the order in which it is executed, in your logic you have it doing a Dial first and then executing MixMonitor. Dial blocks if the call is answered and will stop afterwards, so MixMonitor is never executed. You need to swap the order.

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ok i am making the change the priority change and see if it’s working.

If that doesn’t work then you’ll need to include the Asterisk console output to show what is being executed.

still no file is being produced in the monitor folder. here is the Asterisk console output.

== Using SIP RTP CoS mark 5
> 0x3341e30 – Strict RTP learning after remote address set to: 192.168.179.1:40038
– Executing [03111308308@my-sip:1] MixMonitor(“SIP/4000-00000065”, “.wav, b”)
– Executing [03111308308@my-sip:2] Dial(“SIP/4000-00000065”, “SIP/03111308308,30,Ttr”)
== Using SIP RTP CoS mark 5
== Begin MixMonitor Recording SIP/4000-00000065
– Called SIP/03111308308
– SIP/03111308308-00000066 is ringing
> 0x7f3f0c00b9f0 – Strict RTP learning after remote address set to: 192.168.179.1:6058
– SIP/03111308308-00000066 answered SIP/4000-00000065
– Channel SIP/03111308308-00000066 joined ‘simple_bridge’ basic-bridge <69b84556-8455-41f5-b6a4-04044ca84edf>
– Channel SIP/4000-00000065 joined ‘simple_bridge’ basic-bridge <69b84556-8455-41f5-b6a4-04044ca84edf>
> 0x7f3f0c00b9f0 – Strict RTP switching to RTP target address 192.168.179.1:6058 as source
> 0x3341e30 – Strict RTP switching to RTP target address 192.168.179.1:40038 as source
> 0x3341e30 – Strict RTP learning complete - Locking on source address 192.168.179.1:40038
– Channel SIP/4000-00000065 left ‘simple_bridge’ basic-bridge <69b84556-8455-41f5-b6a4-04044ca84edf>
– Channel SIP/03111308308-00000066 left ‘simple_bridge’ basic-bridge <69b84556-8455-41f5-b6a4-04044ca84edf>
== Spawn extension (my-sip, 03111308308, 2) exited non-zero on ‘SIP/4000-00000065’
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/4000-00000065

Have you set a recording dialplan variable? It appears not as it is recording to “.wav”

how do you do that? Please excuse me i am new to asterisk. I used to write similar statement before and two files gets produced. one for the IVR and other was actual call recording.

Please help me with setting a recording dialplan variable. i will be very thank full.

If you don’t know how to set a dialplan variable you are missing fundamental dialplan logic knowledge, which will hurt you if you need to do anything further including figuring out why things don’t work (such as this). I would HIGHLY suggest you read documentation online such as the Asterisk book or else you will run into problems in the future and be at a loss as to what to do.

The book itself has a part on this[1].

[1] http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#asterisk-DP-Basics-SECT-3.5

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Thank you sir for taking out the time to help me. I really appreciate your help. I will read the book now. Thanks a lot