similar to: How to count ongoing calls from the dialplan

2008/11/4 Igor Zamocky <asterix at ponozky.sk> > > Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers > ? > I didn't. Now I did and it's working the way I wanted. Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and SIPPEER but limitonpeers is much more concise. Thanks a lot. > > > Hi, > > > In this

Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

Hello, In sip.conf, each peer/friend/user entry gathers several parameters such as type, canreinvite or mailbox. How can you specifically access to mailbox value from dialplan ? I know how to access custom parameters (ie setvar=FOO=value) but I don't know to access standard parameters. I'm specifically concerned to access to mailbox's value (from a given entry) but would be

I have been doing some reading about gtalk and asterisk. Most of it is pointed to enable using gtalk for making phonecalls. Would it be possible to use gtalk instant messaging input (just some text send to the gtalk account configured on an asterisk box) into the dialplan. This way you could use gtalk im to trigger all kind of events like sending an sms, adding sip entries to the system,

Hi, I know you can access various sip variables via 'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of the sip user - but what about variables? I have a user that has setvar=123456 in their users.conf (sip.conf if you prefer). I can read it with a 'sip show peer 201' - but that gives everything and parsing that isn't really an option. Anyone know how

2009/3/4 Klaus Darilion <klaus.mailinglists at pernau.at> > core show function SIPPEER Thanks : that's exactly what I was looking for !! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/7d05f172/attachment.htm

Hi gang, Since I'm getting no joy from device_Status or SIPPEER in 1.4.26-rc1, I thought I would do an AGI to read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet;

Hi list I plan to implement a prepaid solution where the system needs to check for remaining credit periodically during a call. The reason for this is that this is a system where the credit pool can be used simultaneously by more people, and not only for calling. I have a problem figuring how to be able to run logic while a call is in progress. The L(x:y:z) option to Dial() is good, but not

Hi, What is currently missing in Asterisk ecosystem to get 2 servers active-active redundancy such as when server 1 is failing (in some circumstances), its ongoing calls (or most of them) are kept alive and handed over to server 2 ? I remember that a couple of years ago, Avaya claimed it could achieve this with its high end servers. Could it be possible with Asterisk ? Will SCF change this ?

Hi, In sip.conf, it's possible to add a line such as setvar=MYFIELD=foo and access this value from diaplan with SIPPEER function. 1. Which function is available to access values in users.conf such as vmsecret ? 2. Is it possible to extend users.conf with custom keys/values ? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, With setvar statements in chan_dahdi.conf, we have a convenient way to store DAHDI channels specific values. Unfortunately, we don't have a function to access this data from the dialplan as easily as SIPPEER ou IAXPEER would for SIP or IAX trunks. Using AST_CONFIG, you can access DAHDI setvar value but: 1. only one setvar value (see bellow) 2. AST_CONFIG reads values from current

On 06.11.2015 14:06, Rusty Newton wrote: > > It should be able to handle typical regular expression. I don't see > anything wrong with what you are doing. Please file a bug at > issues.asterisk.org/jira. Do include a debug log on the issue > captured when Asterisk attempts to execute these extensions. > >

Hi, How can you add specific statements into Asterisk dialplan (extension.ael, ...) for attented transfers ? I can see Asterisk sending Transfer or Masquerade events through AMI (in 1.6.1) but I could use an external program to catch those events but I would prefer to use dialplan instead. Any idea ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Summary: Is it possible to append to a user-defined text file from the dialplan without using AGI ? Background: We are using our Asterisk server to place outbound calls alone. We've 8 E1s that enables us to place about 240 calls concurrently. The server is a bit old, and we are not able to reach our maximum capacity. The dialplan makes use of MySQL to record call status information to our

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

Hi, I'm stuck on a silly thing. I need to get the "billsec" CDR value after a call. But I'm finding its always 0. Here's my test code: exten => *244*,1,Dial(Local/test@custom-tests/n,,g) exten => *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is ${CDR(billsec)}) exten => *244*,n,Hangup [custom-tests] exten => test,1,Answer exten =>

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

Hallo all! Know somebody, if exist Dialplan commands (specifically sql sentences) for Asterisk Realtime? For example: I have users defined in mysql database. In the dialplan, I would like to select one field of a table. select email from sip_buddies where name=200 I try to use DBget, but I have error. I think because DBget use intern Database, and can 't connect to mysql. (Sintaxis: