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[asterisk-users] R: Asterisk and Call Hold


 
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mcolombo at enter.it
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PostPosted: Wed Jul 16, 2014 10:39 am    Post subject: [asterisk-users] R: Asterisk and Call Hold Reply with quote

Hi All,
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”.
I’ve already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem.
I’ve already read all the information about canreinvite and directmedia

Can anybody help me?

Thanks a lot
Marco
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jcolp at digium.com
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PostPosted: Wed Jul 16, 2014 11:02 am    Post subject: [asterisk-users] R: Asterisk and Call Hold Reply with quote

Marco Colombo wrote:
Quote:
Hi All,

Kia ora,

Quote:
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

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