similar to: Problems with ChanIsAvail always return status 0

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten =>

Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch Sendt: 27. juni 2006 09:10 Til:

My teeth are on edge after this one. A couple of perfectly good hours of my life, and I still don't know what's going on. . . . The extensions.conf.sample that comes with the current SVN trunk has this line, in an example that shows how to use ChanIsAvail: exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) I couldn't get this to work unless I surrounded the

Hi. I'm currently setting up a particular conference: 3 members (a,b,c), a can speak with b and c, b and c can speak only with a and not between them. I found my possible solution with paging/intercom using option "d" (full-duplex), but I need to make ringing the phone in intercom. Now I set auto-answer=6 but after first ring the phone hangup the call. There is a way to using

Hello list. ChanIsAvail returns 20 for ${AVAILSTATUS}. What does this '20' mean ?? ... exten => 1,n,ChanIsAvail(SIP/sin10) exten => 1,n,NoOp(chanisavail == ${AVAILSTATUS}) ... [Mar 4 15:10:16] -- Executing [1 at sin:7] ChanIsAvail("IAX2/testlocal-14088", "SIP/sin10") in new stack [Mar 4 15:10:16] -- Executing [1 at sin:8]

I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 phones in a paging group. I have all the phones set up with an extra line that auto answers the dial from my paging extension when the primary line is not in use. All of these are operating correctly however the 7961/7970s all ring once and then auto answer so the person paging all the phones has the first part of his

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

Situation: Dialplan requires several phones be rung on an incoming call, some of which have multiple line presentations. If Line 1 on a phone is busy, Line 2 should be rung instead. Problem: Normal Dial() syntax only rings ALL extensions which we do not want since we only want the second line on a given phone to ring if the lower number line is busy. More Detailed example: 3 phones with 2

Has anyone else had problems with the Page() application not working under Asterisk 1.2.3? We use Cisco 7960 phones and set one of the lines to auto answer. When someone dials the paging extension it calls the page app and invites all the lines on the phones that are set to auto answer into a meetme conference where all the members are muted except the original caller. When I try to use the

Hi I am trying to use ChanIsAvail to decide if a particular extension is available in the sip channel I am using MySQL to hold my SIP friends. and wy cvs version shows Asterisk CVS-08/02/04 my intention is, that if the extension is not available in Sip channel, I will send the call somewhere else my extensions file contains the following: exten => _[123]XX,1,ChanIsAvail(sip/${EXTEN}) exten

Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten => s,7,ChanIsAvail(IAX2/anton:intrudercom@armando-gw) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any

hello, Although my previous posts in this forum have not received satisfying answers, here is another question from me. my question is can i use ChanIsAvail function to get the status of a user in the format SPI/user-id if i provide user in sip uri like this ChanIsAvail(SIP/user at 153.18.x.x:5062) calling user with this sip uri works fine. I once tried but status returned was "unknow

Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro. The logic is simple; try Zap/g1 (a group of two E1s), and if that fails, try locating a channel via DUNDi. Here's a massively cut down version to illustrate the problem I'm having. macro dialout ( dest ) { ChanIsAvail(Zap/g1); noop(Value of AVAILCHAN is ${AVAILCHAN});

Curious whether the ChanIsAvail command is thread safe. By that I mean, if I use ChanIsAvail to determine which channel to use, can I be sure that it will still be available when I go to Dial it on the next line? It occurs to me that there's a possibility the channel could get used by a competing thread AFTER my thread has determined it is available and BEFORE my thread gets a chance to

I'm testing ChanIsAvail context and it is not working for me. exten => 55,1,ChanIsAvail(SIP/11&SIP/21) exten => 55,2,Cut(theChannel=AVAILCHAN,,1) exten => 55,3,Dial(${theChannel},r) exten => 55,4,Hangup exten => 55,102,Goto(s,4) It is not dialing SIP/21 when I'm talking on SIP/11, it execute Hangup instruction instruction. According to notes: The channels are checked

Hello all. Has anyone had any success using ChanIsAvail with only SIP channels? Is there another, better way to check if an extension is busy without dialing it? Thanks, B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040109/48ac2c3e/attachment.htm

Hello everybody, i have two gsm line (extra channels) and i'd like to schedule the outgoing calls with a round-robin strategy. If all the gsm lines are busy, the call must be sent to the pri lines with a linear strategy. here is the dialplan: exten => gsm,ChanIsAvail(EXTRA/r2&DAHDI/g1) same => n,GotoIf($["${AVAILORIGCHAN}" = ""]?unavail,1) same =>

I sent this earlier under "Editting variable contents" but no-one has responded. So, the subject is now more to the problem, instead of the solution I was trying to implement. ChanIsAvail returns the channel ID plus "-<session>". How can I edit ${AVAILCHAN} to remove this session ID, so I can use its contents in a subsequent Dial statement? Dialing on Zap just gives a