similar to: How to hangup all channels

Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? For example: extensions.conf =============== [context] exten => _X.,1,MYSQL(Connect connid localhost user pass db) exten => _X.,n,MYSQL(Query resultid ${connid} SELECT `something` FROM `table` WHERE `number` = ${EXTEN}) exten => _X.,n,MYSQL(Fetch foundRow ${resultid} something)

Hi, i have context in a dialplan, I want to "execute" this context without insert the Answer Application (s? ..without call any ext). Example : [sistema-allarmi-principale] exten => s,1,Set(GRUPPO=${DIAL:-2:1}) exten => s,2,Set(ALLARME=${DIAL:1:1}) exten => s,3,AGI(checkgroup.php|${GRUPPO}) ;rest of... I tried with a Call Data File.. i create a CallDataFile like this :

We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian

Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person then puts the caller on-hold. No moh is heard by the caller (in fact, they get silence).

Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like <disclaimer> at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest

Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk

With endbeforehexten=no I actually get two CDR entries. One for the call and a second one for the "h" extension. "","13","10","sip-locals","""13"" <13>","SIP/13-00000006","SIP/10-00000007","Dial","SIP/10","2015-08-04 06:28:44","2015-08-04

Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is "dropped" until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to

So... I'm trying to access CDR(duration) and CDR(billsec) inside h... I keep getting 0. Can I access the CDR function inside a hangup extensions? Asterisk 1.4.13 Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the

Hi, Reading this old thread, may I ask if keeping hangup handlers from updating CDR values still enforced in Asterisk 15 ? If positive, would it be very complex to add in Asterisk, a configuration option allowing a system administrator to list in cdr.conf, the CDR fields allowed to be updated in hangup handlers ? I'm planning to store some RTCP stats. Saving them in CDR(userfield) would be

Problem as follows: [default] exten => 777,1,Gosub(sub,1,1) exten => 777,n,Hangup() exten => h,1,NoOp(hung up in 'default' context) [sub] exten => 1,1,NoOp(in sub) exten => 1,n,Playback(tt-monkeys) exten => 1,n,Return() exten => h,1,NoOp(hung up in 'sub' context) This works fine if the caller listens to all the 'tt-monkeys' and let's the system

Hi, Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can find a link to download a backported for Asterisk 1.4 version of DEVICE_STATE function. (Elsewhere, you can find reference to another backported function DEVSTATE which seems to behave the same as DEVICE_STATE). As I would like to prepare as much as possible, my dialplan to 1.6 and beyond, I would prefer to use

Hi Guys, I have a channel that is dialed with *Timeout* option. So, there is definite time to it. Only thing is that I don't have control of that channel. I only know that it's using g729 codec and that there is only one channel that is using g729 at any given time. So, my question is:

Is it possible to execute lines in a script when the caller disconnects? I.e. to check when the last person leaves a conference room so you know it is safe to move the recording file. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130829/37108347/attachment.htm>

Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call state of a SIP peer ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I haven't found an example on how to compare the value of a string variable with spaces in it, and the While loop below never exits: ========== extensions.conf exten => start,n,Set(MYVAR="Dummy value") exten => start,n,NoOp(${MYVAR}) ;BAD TOO ;exten => start,n,While(!$[${MYVAR} : "Some string"]) exten => start,n,While($[${MYVAR} != "Some

Any news for this problem. Who has this problem Why you don't answer. --- En date de?: Jeu 21.10.10, ALAEDDINE abbech <alasupcom at yahoo.fr> a ?crit?: De: ALAEDDINE abbech <alasupcom at yahoo.fr> Objet: thousands Hangup per second /saturation of bandwidth ?: asterisk-users at lists.digium.com Date: Jeudi 21 octobre 2010, 11h42 Hello, I have a problem of saturation of

Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call

Hello All, one of the extensions fall into a loop, I don't know how to hangup that channel -- Executing [i at autoatten:2] Goto("Local/100 at sipphones-000001b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200 at sipphones-000001b2;2 -- Executing [i at autoatten:1]