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Chanisavail  Checks if some of those asked stations are available. 

Be aware the AVAILSTATUS variable is utilized for both apparatus state and cause code. It’s thus possible in order for it to provide a price that can indicate a gadget is available as it is not. It’s suggested that your AVAILORIGCHAN factor can be used alternatively to see if it’s the gadget is not. 

New in Asterisk 1.8: The ChanIsAvail program now unlocks the Came Back cause code using Another variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS 

The following variables would likely be put via This application: 

$AVAILCHAN — that the title of this available station, in case one proceeds 

$AVAILORIGCHAN — the only real station name which was used to make the station 

$AVAILSTATUS — the status code to its available station (see”device state.c“) 

0 AST_DEVICE_UNKNOWN –“Unknown”; station remains legitimate, but the not known state. 

1 AST_DEVICE_NOT_INUSE –“Maybe not in usage” 

2 AST_DEVICE in-use –“inuse“; station is currently in use. 

3 AST_DEVICE_BUSY –“hectic”; the channel is currently busy. 

The application form returns”either the device condition and source code.” 

Selections (brand new in Asterisk v1.2?) : 

S — Consider the channel inaccessible if the station is still in use in all (buggy with SIP, always returning 0?!) 

J — Confirm jumping to priority n+101 if no station can be found. 


At Present, ChanIsAvail just works with ZAP, IAX2, mISDN, Skinny, and SIP stations. MGCP stations aren’t supported. 

If none of the requested channels are available, the new Asterisk priorities are going to be n+101 (except if this type of priority does not exist). 

If anyone of the request channels is available, then the following priority is going to be +1. The channel factor $AVAILCHAN is likely to soon be set to the name of this available station. 

The channels will be assessed in the order record, coming to the first available channel at the list in $AVAILCHAN. 

Be aware that ChanIsAvail() returns not just the title of this station at $AVAILCHAN, but in addition, appends the station’s SESSIONID. You will probably have to strip the session ID away, as illustrated in the case under. 

ChanIsAvail is maybe not a solution number to inform you whether the channel is not, it is largely to tell you whether it’d be possible to mail a telephone there. Whether that call might wind up being approved or not is solely up to the peer we mail the telephone to, plus so they are able to easily deny the telephone though they usually do not appear to be ‘active.’ 

So: If you would like to use ChanIsAvail to decide if the SIP peer-reviewed is understood and enrolled, it is going to continue to work fine. If you wish to make use of it for limiting simultaneous phone calls to your peer, it’s not going to do the job reliably for you. 

Sip Peers

For notification whether Sip peers are not or online, once you’re employing neglect, then you may want to just use the SipPeer(title’: standing ) function, and jump primarily based on this. ChanIsAvail will not seem to inform you of the difference in between a Sip peer-to-peer that is online, and one who’s offline. Example snippet: 

Extend => s,n,Set(VOIPCHECK=0) 

Extend => s,n,Place (PEERCHECK1=myprovider-out); SIP peer-reviewed name as described in sip.conf 

; Be certain to possess qualifies allowed with this SIP peer! 

In the event that you aren’t using Realtime, you can assess the SIP/Registry tree of AstDB to your peer-reviewed in concerns, such as together with the help of the DB_EXIST functionality. 


; Watch if level two will be available. If not, decide to try lineup 1. 

We, Will, Need to strip the session off ID and Dial’12345678′ at Zap/2. 

; Should neither line 2 nor line Inch can be obtained, play a message 

Mobile Phone

According to insect 4506, Chanisavail is not designed to find if your mobile phone is in use or not whatsoever, it is simply intended to check if asterisk may deliver the call out there. 

I attempted employing call-limit and Chanisavail SMS, but it’s busted in SIP. In use gets implemented only to friends, and when it gets an incoming call that is not answered, it gets decremented and also doesn’t remain exactly the exact same, which will be a pest infestation. 

You need to look at applying bands alternatively. 

Limit incoming calls using DEVICE_STATE and also Asterisk 1.6 

To limit In-Coming simultaneous calls into the peer, strive DEVICE_STATE. 




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