[asterisk-users] R: Asterisk and Call Hold

Joshua Colp jcolp at digium.com
Wed Jul 16 11:01:45 CDT 2014

Marco Colombo wrote:
> Hi All,

Kia ora,

> I have a problem with asterisk and call hold.
> In the re-invite package when I take the call to the hold, the SDP value
> “a=sendrecv” is present, according to the rfc3264 the sdp value a must
> be mark with “sendonly”.

Are you referring to a call being put on hold? If so this is correct. 
Internally the musiconhold just becomes a different source of audio, the 
fact it is on hold does not get reflected out the SIP signaling.

People have mentioned they'd like this (as well as being able to 
passthrough a hold request) but nobody I know of has worked on it.


Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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