I am using asterisk between Microsoft Speech Server and a hardware PSTN Gateway for redirecting standard calls to international phone users.
We are using asterisk because we needed to implement a free message before answering the call.
The solution seemed working fine, but we found that when the call is estabilished (ie the call arrived, the asterisk played the message, then connected to the Microsoft Speech server and the call is tranfered to the new number), the asterisk process stops without any reason, causing the chain of SIP messages to interrupt and the service is unavailable.
The version we are using is Asterisk 18.104.22.168 on Linux CentOS 5.2 with Kernel 2.6.18
Can anyone help with this problem.
We asked to an expert of asterisk that said the configuration files seemed correct.
Thanks in advance Andrea Perazzolo