Hi guys: Has anyone noticed a difference in the behaviour of the Dial cmd since somewhere around asterisk 1.2.15.
We use the command Dial(SIP/123456789,24,o) to dial customers where 123456789 is their SIP peer (and phone number) Previously, if the customers VoIP device (ata or phone) was offline or not registered it would immediately consider it a busy call and jump N+101 to the busy call handling routines. This seems to have changed. Now it continues to try and retransmit INVITE messages for 24 seconds (or whatever timeout is in the Dial command), it which point it gives up and treats it as an unavailable and jumps N+1. I would like to have it working the old way again...has anyone else run into this? Regards, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
