<div>
Try Mix:1 instead of Mix:true.
</div>


<div>
I tried that as well. Same result.
</div>


<div>
Anyone with Asterisk 1.6.2.14 who can try this?
</div>


Anyone with Asterisk 1.6.2.14 who can try this?

<div>
<div class="content wysiwyg-content">
Hi guys.<br />
I am using the asterisk manager to record calls on my asterisk 1.6.2.14. The action on the manager interface goes like this:<br />
<br />
Action: Monitor<br />
Channel: &lt;channelname&gt;<br />
Mix: true<br />
<br />
This is executed with success and the recording starts on the specified channel and the bridged channel as well. When the call ends asterisk puts the two wav-files into one and here the problem occurs. The wav-file consists of first the voice from one channel and then the voice from the bridged channel. They should be mixed together, not player after each other. If I set Mix: false the result is exactly the same. I have done this many times in Asterisk 1.4 without issues. What can be wrong here?<br />
<br />
Thanks
</div>
</div>


Hi guys.
I am using the asterisk manager to record calls on my asterisk 1.6.2.14. The action on the manager interface goes like this:

Action: Monitor
Channel: <channelname>
Mix: true

This is executed with success and the recording starts on the specified channel and the bridged channel as well. When the call ends asterisk puts the two wav-files into one and here the problem occurs. The wav-file consists of first the voice from one channel and then the voice from the bridged channel. They should be mixed together, not player after each other. If I set Mix: false the result is exactly the same. I have done this many times in Asterisk 1.4 without issues. What can be wrong here?

Thanks

Asterisk: AMI Monitor action - Mix fails

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

IP Telephony

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).