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.
Cheers, -- 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
