On 6/20/10 5:08 PM, Iñaki Baz Castillo wrote: > 2010/6/20 Philip Prindeville<[email protected]>: > >> (2) When the call rings out to the cell phone, is there a header that gets >> translated into SS7 that tells cell carrier being rung to not do a >> forward-on-no-answer to voicemail? In the above scenario, we want the >> voicemail on the PBX to answer, not the voicemail on the cell's carrier to >> pick up if there's no answer. >> >> >> There is no such specification. Voicemail serves are not a "standard", >> they are just servers that automatically answer the call to bill it :) >> >> >> Right, but forwarding from the handset to the voicemail IP (intelligence >> peripheral) counts as an additional forwarding step. >> > No, in SIP it would be a 302 which is a SIP response rather than a new step. > If the called phone would *route* the INVITE back to the proxy (by > changing the RURI) then it would be a new step, but this is not what > happens as phones are UA's rather than proxies. > > > >> If one sent a header saying Max-Forwards N, with a current Hop-Count as N, >> then that would be sufficient to stop the additional transfer, yes? >> > No. First of all I don't know if there is such a "Max-Forwards" header > out of SIP world (as you are speaking about SS7). Anyhow, considering > it exists: > > - Phone1 calls Phone2 through Proxy (max-forwards=1). > - Proxy decrements max-forwards to 0 and routes the call to Phone2. > - Phone2 doesn't answer in N seconds,or it switched off, or rejects > the call by sending a 302 to redirect the call to its voicemail. > - Proxy then decides to create a new branch (***so Max-Forwards > remains the same***) to the voicemail server. > > There is no way at all to instruct the PSTN provider not to forward a > call to the voicemail as such decision just depends on the called > number's configuration and provider's configuration. > > >
Yeah, it's been a while since I've had to look at SS7: last time was working on F-Node functionality at Cisco in 2002... How much interaction does the SIP community have with the SS7 standards group and SS7 implementation? I'm thinking that it would be useful to have both SS7 and SIP have an associative directive or information element that controls forward-on-no-answer and forward-on-busy to voicemail. Even if the call originator can't control it, the remote party could at least generate a progression indication that would allow the caller to terminate the call at that point. Of course, due to latency in the network, that might need to be done by the closest SIP proxy, rather than all the way back at the originator. -Philip _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
