Subject: Re: How to extract sip-id when using alias-id? - msg#00433

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Re: AVP_RADIUS Question

Ricardo, yo may find this helpful...it's similar topic http://openser.org/pipermail/users/2006-April/004225.html regards, bogdan Ricardo Martinez wrote: Hello list. If i need to get several AVPs from the Radius answer message, how is the format for ATTRIBUTE SIP-AVP in the Radius message?. I need 3 AVP's from a User : state1:off state2:on state3:sip:12345-3Q2Tfjf0mexWk0Htik3J/w@xxxxxxxxxxxxxxxx When i need just one AVP to load is easy to add the Radius parameter SIP-AVP like this SIP-AVP=state1:off But to add the other two AVP's how is the format ? SIP-AVP=state1:off;state2:on;state3:sip:12345-3Q2Tfjf0mexWk0Htik3J/w@xxxxxxxxxxxxxxxx ???? Can someone help me here ? Thanks in advance Regards, Ricardo Martinez.- _______________________________________________ Users mailing list Users-WLQjAxnOB31AfugRpC6u6w@xxxxxxxxxxxxxxxx http://openser.org/cgi-bin/mailman/listinfo/users

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Re: Zombie/hanging calls - TCPKeepalive timeout possible ?

Hi Gerry, no, there is no such mechanism available since it's not reliable - for example, during a call, more TCP connections may be used - one for the INVITE, close the connection during the call and open a new one for BYE - it's perfect possible, so you cannot rely on the transport status. there are other ways to do it - at signalling level, via Session Timer; at media level - see when no more RTP traffic comes form one point. regards, bogdan Gerry wrote: Hi, Is there any functionality implemented in SER/Openser which allows to detect dead endpoints ? Gnugk implements this with tcpkeepalive timeouts. Is there any equivalent or something similar in SER which would prevent non-active calls to hang - and incurr possibly large costs ? TIA for your help in the matter Gerry ___________________________________________________________ NEW Yahoo! Cars - sell your car and browse thousands of new and used cars online! http://uk.cars.yahoo.com/ _______________________________________________ Users mailing list Users-WLQjAxnOB31AfugRpC6u6w@xxxxxxxxxxxxxxxx http://openser.org/cgi-bin/mailman/listinfo/users

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How to extract sip-id when using alias-id?

I am using alias_db to store alias's for users. However when an alias is dialed I need to also be able to restore the alias owners original sip-id. This is necessary to forward the call to voicemail using sip-id. Any guidance on getting the sip-id out of the alias_db and replacing the alias-id prior to forwarding to vmail? Thanks in advance! F __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: How to extract sip-id when using alias-id?

Good point. I need to clarify... The alias is working to route to the user however if the timer expires, the route fails or if there is a reject, the failure routes are applying their logic to the alias-id not the sip-id. What I need to do is apply the failure route logic to the sip-id so we can send the attempt to voicemail. Let say I have an alias of 1000-9dM3hzoSjHY@xxxxxxxxxxxxxxxx for sip-id 5551212-RsIHbiFnPJY@xxxxxxxxxxxxxxxx When 1000-9dM3hzoSjHY@xxxxxxxxxxxxxxxx is dialed by another user in the domain (I am using "use_domain") the call is properly routed to: 5551212-RsIHbiFnPJY@xxxxxxxxxxxxxxxx However, if 5551212-9dM3hzoSjHY@xxxxxxxxxxxxxxxx does not answer, timer expires, or a reject, then the failure logic is be applied to the alias. Failure route says: prefix("66"); append_branch(); append_urihf("CC-Diversion: ", "\r\n"); t_relay(); Result of the failure logic: 661000-9dM3hzoSjHY@xxxxxxxxxxxxxxxx Needs to be: 665551212-9dM3hzoSjHY@xxxxxxxxxxxxxxxx Thoughts? F --- Bogdan-Andrei Iancu <bogdan-/qk4c3+6r1L0ABVSK+4LOw@xxxxxxxxxxxxxxxx> wrote: > Hi, > > not sure I understand - you dial an alias and via > db_aliases you replace > the alias with the sip-id. So you have the sip-id .. > where is the problem? > > regards, > bogdan > > Frogger wrote: > > >I am using alias_db to store alias's for users. > > > >However when an alias is dialed I need to also be > able > >to restore the alias owners original sip-id. > > > >This is necessary to forward the call to voicemail > >using sip-id. > > > >Any guidance on getting the sip-id out of the > alias_db > >and replacing the alias-id prior to forwarding to > >vmail? > > > >Thanks in advance! > > > >F > > > >__________________________________________________ > >Do You Yahoo!? > >Tired of spam? Yahoo! Mail has the best spam > protection around > >http://mail.yahoo.com > > > >_______________________________________________ > >Users mailing list > >Users-WLQjAxnOB31AfugRpC6u6w@xxxxxxxxxxxxxxxx > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com