<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I'm using 1.2.6 with the dialplan I posted so I guess the UA you are using is just plain hosing you.<div><br class="webkit-block-placeholder"></div><div>Anyway, with the queue I believe the music on hold is played to the inbound side until the call is picked up by an agent. The queue tries every <retry> seconds to get an agent for <timeout> seconds. If that fails for however long you set the limit to then the dialplan continues. I use this to set the priority of the call a little higher and loop it back into the queue but it could be used for a problem such as yours just as easily. I don't use the agent login and all that so I may be talking about something that doesn't apply to your configuration. If so, sorry for wasting your time.</div><div><br class="webkit-block-placeholder"></div><div><div>exten => +1XXXXXXXXX,1,NoOp(Inbound call from ${CALLERIDNUM})</div><div>exten => +1XXXXXXXXX,n,Answer()</div><div>exten => +1XXXXXXXXX,n,Set(GROUP()=cloud)</div><div>exten => +1XXXXXXXXX,n,Set(QUEUE_PRIO=0)</div><div>exten => +1XXXXXXXXX,n(waiting),Queue(mainline,,,,600)</div><div>exten => +1XXXXXXXXX,n,Set(QUEUE_PRIO=$[${QUEUE_PRIO} + 5])</div><div>exten => +1XXXXXXXXX,n,GoTo(waiting)</div><div>exten => +1XXXXXXXXX,n,HangUp</div><div><br class="webkit-block-placeholder"></div><div><div>On Dec 2, 2007, at 7:02 PM, Vieri wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"> <div> <!-- Converted from text/plain format --><p><font size="2">I'd like to add that "show hints" on * CLI displays<br> the following for ext 4053 tested below:<br> <br> 4053 : SIP/4053 <br> State:Idle Watchers 0<br> <br> (it should be "unavailable" or something, but anyway,<br> ChanIsAvail reports an AVAILSTATUS of 0, ie. unknown)<br> <br> --- Vieri <<a href="mailto:rentorbuy@yahoo.com">rentorbuy@yahoo.com</a>> wrote:<br> <br> > Thanks Richard but I think that ChanIsAvail must be<br> > buggy (based on some user comments in the wiki,<br> > although quite outdated).<br> ><br> > I have the hint entry as you say (am using FreePBX<br> > and<br> > it's already there).<br> ><br> > But whenever I call ChanIsAvail with the s option I<br> > always get:<br> > ${AVAILSTATUS} = 0 AST_DEVICE_UNKNOWN - "Unknown";<br> > channel is valid, but unknown state.<br> ><br> > I might be doing something wrong but here is the<br> > code:<br> ><br> > [IVR-menu1]<br> > exten => s,1,Answer()<br> > (...)<br> > exten => s,n,Playback(welcome)<br> > exten => s,n,ChanIsAvail(SIP/4053|s)<br> > exten => s,n,NoOp(DEBUG: availstatus is<br> > ${AVAILSTATUS})<br> ><br> > In extensions.conf I also have:<br> > exten => 4053,hint,SIP/4053<br> ><br> > I'm using Astrisk 1.2. Is ChanIsAvail working well<br> > in<br> > 1.2?<br> ><br> > As far as setting a time limit on a call in the<br> > queue<br> > is concerned, it doesn't sound "nice" for the caller<br> > to be dropped after a few rings when it could have<br> > been dropped right fom the beginning. It could be a<br> > solution but it doesn't sound "right" ;-).<br> ><br> > Vieri<br> ><br> > --- Richard Revels <<a href="mailto:rrevels@bandwidth.com">rrevels@bandwidth.com</a>> wrote:<br> ><br> > > In the sip.conf entry assign a context.<br> > ><br> > > In that context, hint the extension i.e. exten =><br> > > 7302,hint,SIP/7302.<br> > ><br> > > Before you get ready to dial, or whatever, do<br> > > chanisavail i.e.<br> > ><br> > > exten =><br> > > _1XXXX,n(CheckUse),ChanIsAvail(SIP/${EXTEN:1},js)<br> > > exten => _1XXXX,n,Playback(beep)<br> > > exten => _1XXXX,n,Dial(SIP/${EXTEN},2)<br> > > exten =><br> > > _1XXXX,n,Goto(result-${DIALSTATUS},${EXTEN},1)<br> > > exten => _1XXXX,CheckUse+101,SayDigits(${EXTEN:1})<br> > > exten =><br> > _1XXXX,CheckUse+102,Playback(vm-isonphone)<br> > > exten => _1XXXX,CheckUse+103,Hangup()<br> > ><br> > > This is from the paging stuff. It checks the<br> > > primary extension before <br> > > ringing the auto answer extension of the phone. I<br> > > seem to remember it <br> > > detecting DND as well for the Cisco 7960.<br> > ><br> > > I don't see it in this message but I seem to<br> > > remember seeing somewhere <br> > > in this thread that the goal is to keep people<br> > from<br> > > being in a queue <br> > > forever. Why not just set a time limit on the<br> > queue<br> > > and play back <br> > > "all operators busy" and hang up if a call hits<br> > that<br> > > limit?<br> > ><br> > > Richard<br> > ><br> > ><br> > ><br> > > On Dec 2, 2007, at 8:51 AM, Vieri wrote:<br> > ><br> > > > Hi,<br> > > ><br> > > > I am trying to get a SIP extension's status<br> > > without<br> > > > actually making a call.<br> > > ><br> > > > I am using sofia-sip's "options" example utility<br> > > and<br> > > > the sip clients are SJphone softphones.<br> > > ><br> > > > From Asterisk I run the "options" utility and<br> > > query a<br> > > > sip extension at 10.215.147.240. I get:<br> > > ><br> > > > # ./options -1 --all <a href="sip:10.215.147.240">sip:10.215.147.240</a><br> > > > SIP/2.0 501 Not Implemented<br> > > > Via: SIP/2.0/UDP<br> > > ><br> > ><br> ><br> 10.215.144.27:38098;branch=z9hG4bKUKS02S3F8H8ZS;received=10.215.144.27<br> > > > From: <<a href="sip:10.215.144.27">sip:10.215.144.27</a>>;tag=U3DKgF7HgFKXH<br> > > > To: "unknown" <<a href="sip:10.215.147.240">sip:10.215.147.240</a>>;tag=614733430<br> > > > Call-ID: b6968197-1b7d-122b-0ab0-00c09f10e472<br> > > > CSeq: 92182805 OPTIONS<br> > > > Content-Length: 0<br> > > > Server: SJphone/1.65.377a (SJ Labs)<br> > > ><br> > > > I guess that the softphone should be answering<br> > > with a<br> > > > 2xx code followed by a status description?<br> > > > So I tried with the INVITE method and set DND on<br> > > the<br> > > > SIP extension:<br> > > ><br> > > > # ./options -1 --all --method INVITE<br> > > > <a href="sip:10.215.147.240">sip:10.215.147.240</a><br> > > > SIP/2.0 486 Busy Here<br> > > > Via: SIP/2.0/UDP<br> > > ><br> > ><br> ><br> 10.215.144.27:38098;branch=z9hG4bK5Z3BS8F737t0e;received=10.215.144.27<br> > > > From: <<a href="sip:10.215.144.27">sip:10.215.144.27</a>>;tag=590Z1ND8B6XpN<br> > > > To: "unknown"<br> > <<a href="sip:10.215.147.240">sip:10.215.147.240</a>>;tag=1a2d77b524<br> > > > Call-ID: 668ad4fa-1b7e-122b-fcb6-00c09f10e472<br> > > > CSeq: 92182952 INVITE<br> > > > Content-Length: 0<br> > > > Server: SJphone/1.65.377a (SJ Labs)<br> > > ><br> > > > The above would suit me fine because I get a<br> > "486<br> > > Busy<br> > > > Here" response.<br> > > > However, if DND is off then I get:<br> > > ><br> > > > # ./options -1 --all --method INVITE<br> > > > <a href="sip:10.215.147.240">sip:10.215.147.240</a><br> > > > SIP/2.0 180 Ringing<br> > > ><br> > > > and the SIP extension actually "rings", as<br> > > > expected.(but this is undesireable)<br> > > ><br> > > > Now, does someone know another way to get the<br> > > status<br> > > > (ie. does it accept calls or not?) without<br> > making<br> > > the<br> > > > extension "ring"?<br> > > ><br> > > > Thanks<br> > > ><br> > > > Vieri<br> <br> <br> <br> ____________________________________________________________________________________<br> Never miss a thing. Make Yahoo your home page.<br> <a href="http://www.yahoo.com/r/hs">http://www.yahoo.com/r/hs</a><br> <br> _______________________________________________<br> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br> <br> asterisk-users mailing list<br> To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br> </font> </p> </div> </blockquote></div><br></div></body></html>