I don't know about the 1.4 source, but in 1.2 I guess you would have to add some more code to
handle_response_peerpoke() to handle the case where you got a 486 response from the peer. > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Vieri > Sent: 13 September 2007 18:56 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] how to determine if a SIP > extension has DNDonoroff > > > --- Steve Langstaff <[EMAIL PROTECTED]> wrote: > > > Can you hook into the "qualify" code somehow? - that uses > SIP OPTIONS. > > I already knew of this wiki page: > http://www.voip-info.org/wiki/view/Asterisk+sip+qualify > > So I did a "sip show peer" on the asterisk cli which I am > supposing is the same as the SIPPEER function. > > When SIP softphone has DND turned OFF: > > INF-VOIP*CLI> sip show peer 4053 > INF-VOIP*CLI> > > * Name : 4053 > Secret : <Set> > MD5Secret : <Not set> > Context : from-internal > Subscr.Cont. : <Not set> > Language : es > AMA flags : Unknown > CallingPres : Presentation Allowed, Not Screened > Callgroup : 1 > Pickupgroup : 1 > Mailbox : [EMAIL PROTECTED] > VM Extension : asterisk > LastMsgsSent : 0/0 > Call limit : 0 > Dynamic : Yes > Callerid : "device" <4053> > Expire : 58 > Insecure : no > Nat : Always > ACL : No > CanReinvite : No > PromiscRedir : No > User=Phone : No > Trust RPID : No > Send RPID : No > DTMFmode : rfc2833 > LastMsg : 0 > ToHost : > Addr->IP : 10.215.147.240 Port 5060 > Defaddr->IP : 0.0.0.0 Port 5060 > Def. Username: 4053 > SIP Options : (none) > Codecs : 0xc (ulaw|alaw) > Codec Order : (ulaw,alaw) > Status : OK (127 ms) > Useragent : SJphone/1.65.377a (SJ Labs) > Reg. Contact : sip:[EMAIL PROTECTED] > > When SIP softphone has DND turned ON: > > INF-VOIP*CLI> sip show peer 4053 > INF-VOIP*CLI> > > * Name : 4053 > Secret : <Set> > MD5Secret : <Not set> > Context : from-internal > Subscr.Cont. : <Not set> > Language : es > AMA flags : Unknown > CallingPres : Presentation Allowed, Not Screened > Callgroup : 1 > Pickupgroup : 1 > Mailbox : [EMAIL PROTECTED] > VM Extension : asterisk > LastMsgsSent : 0/0 > Call limit : 0 > Dynamic : Yes > Callerid : "device" <4053> > Expire : 45 > Insecure : no > Nat : Always > ACL : No > CanReinvite : No > PromiscRedir : No > User=Phone : No > Trust RPID : No > Send RPID : No > DTMFmode : rfc2833 > LastMsg : 0 > ToHost : > Addr->IP : 10.215.147.240 Port 5060 > Defaddr->IP : 0.0.0.0 Port 5060 > Def. Username: 4053 > SIP Options : (none) > Codecs : 0xc (ulaw|alaw) > Codec Order : (ulaw,alaw) > Status : OK (127 ms) > Useragent : SJphone/1.65.377a (SJ Labs) > Reg. Contact : sip:[EMAIL PROTECTED] INF-VOIP*CLI> > > I don't see any difference and "SIP Options : (none)" > doesn't look "good". > > (the SIP extension has qualify=yes) > > > > > ______________________________________________________________ > ______________________ > Shape Yahoo! in your own image. Join our Network Research > Panel today! > http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 > > > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --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 > _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
