Hi Tony I can see where you guys are coming from on this and have already enumerated your argument in my own email.
But there are very real reasons for a PBX to signal the hold even when it wants to send its own MOH: 1. Bandwidth: under your scheme the PBX would continue to receive bandwidth-consuming media without using it. 2. Privacy: the far-end has an expectation of privacy while on hold and should have the option to mute automatically when held. 3. Feature richness: signalling the hold enables such innovative features such as reverse hold. 4. ISDN interworking: ISDN supports this and SIP should be compatible with that (as per standard ITU-T Q.1912.5) Also, can you explain why the PBX would use a=sendonly but not dispatch media. Why not a=inactive for that case? > IMHO, PBX-A would be broken if it passed this along the Hold message to > downstream and then started servicing the MOH itself Remember it is not a hold message, it is a media attribute and we are discussing how that should be interpreted within the context of the hold feature in traditional telephony. I would also like to point out in my defence that there are several telephone systems in the field which behave as I described (Nortel BCM50, Aastra Intelligate, Mitel 3300 to name a few). Regards, Richard > I have to agree with Kevin on this one. > > I fail to understand how you have a PBX-A talking to Asterisk talking to > PBX-B and the PBX-A placing the call on hold. Typically you should have a > Client/Phone to PBX-A to Asterisk to PBX-B to Client/Phone/VoiceMail. > > If the Client signals Hold, the PBX should NOT be passing that Hold status on > but transition audio stream from Client to MOH (assuming MOH is handled). > Asterisk shouldn't notice a thing except more RTP packets (or less if it is > my teenage daughter on the phone as the case may be). > > IMHO, PBX-A would be broken if it passed this along the Hold message to > downstream and then started servicing the MOH itself on the RTP stream. That > just doesn't make sense. > > Now if PBX-A were not a PBX and were a SIP Router, and the SIP Router was > attempting this, I can see how it would Re-Invite, but it shouldn't pass the > hold status onto Asterisk. > > Need some clarity here. > > Tony Plack _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
