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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, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged). But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify. Is there any setup or asterisk configuration I need to enable to have A close its call ? Note: when A is already talking with B,

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 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 all I have been looking at my asterisk CDR in the mysql database and some channel names are set to "**Unknown**" string. When I look at the code, everybody when calling ast_channel_alloc set a channel format like SIP/%s or Zap/%s Only app_voicemail.c doesn't when sending emails and I don't use voicemail. Why app_voicemail needs to allocate a channel to send emails ? And in

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 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 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 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