similar to: Cancel a ringing SIP call when the other party disconnect

Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the

Hi all, I need to build an application that will be an SIP server program that will run on Linux and Windows. The sip server need only some features such as be able to : - Register sip endpoints - Answer a call and play a local file - Make a dial from one channel to another. I know asterisk can be stripped to exactly fit my needs. I would like to know if there

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

Hi, I setup an MeetMe conference. So, the admin user calls and enter the conference in talk/listen mode. (Options : dAaxs) Then other users call the same conference and enters in muted mode (options: dlmx) How can the admin user decide, when he is ready to let everybody speaks ? I didn't find such option in the admin menu. Thanks -------------- next part -------------- An HTML

Hi All I'm having some serious kernel panic while using digium cards. It may be related to IRQ shared. Can this cause a lot of drop call and bad voice quality ? Do you guys know if there is a way I can assign one IRQ for each digium card ? Thanks a lot. Here is the output of /var/log/syslog kernel: [ 3821.982893] Uhhuh. NMI received for unknown reason 20. kernel: [

Hi All I want to be able to read some sip informations (from a database) like username, password, host and extension number and place a Dial from asterisk. So basicly, I want to dial sip extensions without modifying sip.conf each time. I don't know, in the dialplan, what the dial string should look like. I tried SIP/<username>:<password>@<host>/<exten>

Hi, I've been debuging the call disconnection problem in our architecture: PSTN---E1---OldPBX---E1---Asterisk This is our problem: -SIP user agent "A" calls a pstn phone "B". -"B" hangs up the call. -SIP user agent "A" starts listenning busytones... But the call still on. (and being payed). - Call only ends when it is correctly hanged up in the

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

Hi, I use asterisk with sip3000 device with "sip-aho" connected to PSTN and "sip-ahi" connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten => 99,n,Dial(SIP/sip-ahi,30,g) exten => 99,n,Hangup() The asterisk properly detects hangup of the caller as I

Hello, Q/ Is it possible to create a DBMS connection automatically on startup of R? (Making sure of course that the db server has been started...) I am running MySQL on Mac OS X 10.4.2 with R2.4.1. I have tried to write a function using the RMySQL commands (below) and place them in .First of .RProfile: drv <- dbDriver("MySQL") dbcon <- dbConnect(drv, {other parameters present in

Hi, First, thanks in advance. Some useful info: >version platform x86_64-unknown-linux-gnu arch x86_64 os linux-gnu system x86_64, linux-gnu version.string R version 2.15.1 (2012-06-22) I'm trying to use the table() function on a 2 column matrix that has 711 million rows (see below). However, it freezes. If I subset the matrix to be less than or equal

Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and

Hi all, I've asterisk 1.2.5 , and what is happening is this: Sip user agent "A" calls a pstn "phone B" Sip User agent Activates MOH. "B" starts listening. "A" doesn't hangup and just Disconnect Sipoftphone XLITE (exit) "B" stills listenning Music on Hold and "A" has left Asterisk, who hangs the call? only when B hangs...

> > > > I would be grateful if anybody could tell me what I should > > tell Verizon > > in NJ so they would enable "disconnect supervision" for my lines. > > > > Apparently "remote hangup notification" or "disconnect > supervision" or > > "calling party control" is NOT the magic phrase for them. Although >

I hope somebody has seen this before... I'm trying to use a Dial command on a inbound call to ring multiple destinations. The calls come in to me from the provider on IAX2, and one of the destinations I try to ring is a IAX2 to call to my cell phone. When I add the IAX2 destination into the Dial command, the setup I am trying to achieve works (i.e. my Zap, SIP, and cell phone all ring) but

I am new to Asterisk and I'm setting up my first Asterisk box. When recieving inbound SIP calls from sipphone.com my friends would not get the ringing indicator after I did an answer command. I found that after playing a gsm file first the ringing started to work like it did for my pots incoming calls. I created a short virtually silent gsm file and that worked as well if played

Hi all I saw in the CDR stocked in mysql as well as those in the csv file that some time, the src field is the same as the dst field which is the extension. When does it happens. Here, we have 4 dgits extensions and most of the time the dst field is the extension and the src field is the 10 digit customer phone number. Do you know when does this happens ?? Thanks Ruddy Gbaguidi

Hi guys, I'm interconnecting an Asterisk box with a Lucent Definity PBX by means of FXO/FXS ports on a TDM2400 card. Everything works well, except for one little thing. Every now and then somebody (from an Asterisk extension) will call another extension on the Lucent Definity PBX and they hit their voicemail. They caller leaves their message (or not) and hangup, BUT the Lucent sometimes

Hello all, I have an Asterisk PBX with the "Queue Log Analyzer" installed [http://www.micpc.com/qloganalyzer]. On the main menu, there's an option of "CALLS COMPLETED [ALL]" where I can see the completed calls that entered any of the queues and my question is: There's a column that states either "COMPLETECALLER" or "COMPLETEAGENT" and I want

Hello I'm encountering a big problem with usbhid-ups driver : the driver hangs after loading some other modules (stale data message) ! And we can't restart it. We load : i2c-piix4 adm1021 adm9240 lm75 The unloading of the modules doesn't solve the problem. # ps -eaf | grep usb /root 188 2 0 07:29 ? 00:00:00 [ksuspend_usbd] root 996 2 0 07:29 ?