Thanks everybody for their answers. I kind of thought that. I can't put the caller id in the first invite.
I need to accept the call from the switch before the switch starts changing abcd bits for ringing. Is there a way to check the status of user? I suppose I could send out an OPTIONS message... that would at least say if the phone was alive. Thanks again. -Mike --- [EMAIL PROTECTED] wrote: > Send Sip-implementors mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, > visit > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > or, via email, send a message with subject or body > 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it > is more specific > than "Re: Contents of Sip-implementors digest..." > > > Today's Topics: > > 1. Re: Invite (hold) question > ([EMAIL PROTECTED]) > 2. Re: interval between two RINGING responses > (zhang jw) > 3. Re: can username part of uri contain "*#" > (Prithvi) > 4. 183 response (jitha) > 5. Re: Invite (hold) question (Sreenath Kulkarni) > 6. Re: interval between two RINGING responses > (Sreenath Kulkarni) > 7. Re: 183 response (zhang jw) > 8. how to exchange messaging in non-ascii? (ST) > 9. Query for OfferAnswer direction (Tang Xi) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 6 Dec 2006 17:33:03 -0500 > From: [EMAIL PROTECTED] > Subject: Re: [Sip-implementors] Invite (hold) > question > To: [email protected] > Message-ID: > <[EMAIL PROTECTED]> > > From: Mike Dorin <[EMAIL PROTECTED]> > > Is it possible to block a uac from putting a call > on hold without > tearing down the original call? > > What are you trying to accomplish? > > Dale > > > ------------------------------ > > Message: 2 > Date: Thu, 7 Dec 2006 10:03:26 +0800 > From: "zhang jw" <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] interval between two > RINGING responses > To: "M. Rangnathan" <[EMAIL PROTECTED]> > Cc: Paul Kyzivat <[EMAIL PROTECTED]>, > Sip-Implementors > <[email protected]> > Message-ID: > > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; > format=flowed > > Hi, > ICT will not time out if it received any provitional > response, and the proxy > will generate a 100 when an INVITE is received, if > the 100 is droped, the > client will keep sending INVITE so the proxy aware > the lose of response 100 > and send it once again. So i don't see the > necessary of periodic Ring > response. > > On 12/7/06, M. Rangnathan <[EMAIL PROTECTED]> wrote: > > > > Well, if the called party takes a while to pick up > the phone and a final > > response is not recieved at the UAC in that time, > and no inermediate > > responses are received either (because the RINGING > was dropped for > > example), I suppose the client transaction on the > UAC would time out. > > Whereas, it would seem that if the INVITE did get > to the UAS and the > > phone is indeed ringing, it would make sense to > send periodic RINGING > > while the phone is indeed ringing at the UAS. > > > > I suppose one can use provisional reliable > response for that purpose if > > that is really an issue but I was wondering why a > single RINGING > > suffices and if so, what is the purpose of even > sending that one RINGING > > if it can be dropped. > > > > Ranga > > > > Paul Kyzivat wrote: > > > > > What do you expect the UAC to do differently > when it receives > > > subsequent 180s that it wouldn't do if it didn't > get them? > > > > > > Paul > > > > > > M. Rangnathan wrote: > > > > > >> Hello > > >> > > >> I want to make a UAS send periodic RINGING > responses as the phone is > > >> being alerted. Are there any guidelines for > what the interval between > > >> such responses should be ? > > >> > > >> Thanks > > >> > > >> Ranga > > >> > > > > > > > > > -- > > M. Ranganathan > > > > Advanced Networking Technologies Division, > > National Institute of Standards and Technology > (NIST), > > 100 Bureau Drive, Stop 8920, Gaithersburg, MD > 20899. > > tel:301 975 3664 , fax:301 590 0932 > http://w3.antd.nist.gov/ > > Advanced Networking Technologies For the People! > > > > > > _______________________________________________ > > Sip-implementors mailing list > > [email protected] > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > ------------------------------ > > Message: 3 > Date: Thu, 07 Dec 2006 05:56:41 +0000 > From: Prithvi <[EMAIL PROTECTED]> > Subject: Re: [Sip-implementors] can username part of > uri contain "*#" > Cc: sip-implementors > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; > format=flowed > > Dear Divya > > As per RFC 3261 section 25.1 > > <RFC3261>------------------------------------------------------------------ > SIP-URI = "sip:" [ userinfo ] hostport > uri-parameters [ headers ] > userinfo = ( user / telephone-subscriber ) > [ ":" password ] "@" > user = 1*( unreserved / escaped / > user-unreserved ) > user-unreserved = "&" / "=" / "+" / "$" / "," / > ";" / "?" / "/" > unreserved = alphanum / mark > mark = "-" / "_" / "." / "!" / "~" / > "*" / "'" > </RFC3261>----------------------------------------------------------------- > > and also please note page no 223 > > <RFC3261>------------------------------------------------------------------ > The BNF for telephone-subscriber can be found in > RFC 2806 [9]. Note, > however, that any characters allowed there that > are not allowed in > the user part of the SIP URI MUST be escaped. > </RFC3261>----------------------------------------------------------------- > > Hence apart from the user part what ever the > character === message truncated === __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
