And in the situation when my client SIP has your computer cut of the network. In this situation software SIP client never sent "BYE".
How can I know when the call SIP already finish?
In H.323, the gatekeeper could detect the end of RTP.
On Tue, 2 Aug 2005, Juan Priotti wrote:
You can execute a php scrip or any other in this way: loadmodule "/usr/local/lib/ser/modules/exec.so"
if (method == "BYE") {
exec_msg("/tmp/putCostCdr.php $SIP_HF_CALLID");
}
so when the BYE is received by SER, the php script is executed, receives the callid and it can rates the call or do whatever you want it to do.
Regards,
Juan Priotti
----- Original Message ----- From: "Daryl Sanders" daryl.sanders@gmail.com To: "Anderson Alves de Albuquerque" anderson@belem.voip.nce.ufrj.br Cc: "SER Users" serusers@lists.iptel.org Sent: Tuesday, August 02, 2005 1:47 PM Subject: Re: [Serusers] Start and fnish call
You could write a script to run periodically that check your acc table for BYEs, then execute your script when it finds a new one.
I think there is a module to that lets SER run an external script from ser.cfg. You might be able to execute this when ser.cfg sees a BYE. I have never used the module, so I'm not really sure. Hope this helps.
- Daryl
On 8/2/05, Anderson Alves de Albuquerque anderson@belem.voip.nce.ufrj.br wrote:
With SER how can I know when the client SIP finish a call?
I need that SER can start a script when the client SIP finish a call.
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