Have you given any thought to moving to at least a current supported version 13? Asterisk 11 has been EOL for some time now I doubt you will get a resolution to a version no longer supported. Moving to the latest version 13 should be relatively quick and painless, and if the issue persists you might find more assistance.
John Novack Andrew Martin wrote:
Hello, I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog POTS lines coming into my Asterisk server from the phone company. Internally, I have about 180 SIP clients defined in sip.conf. What appears to be happening is that if existing calls are consuming all 8 external lines and a new SIP client attempts to make a call, an existing call gets dropped. The asterisk log simply shows this as a normal hangup, so I am not able to easily distinguish between a normal hangup and this type of dropped call. In testing, I am able to get a new SIP client to report "service unavailable" when all 8 lines are consumed, yet still drops are reported. I have been unable to find any configuration settings pertaining to prioritizing existing calls over new calls. What else can I look for to attempt to debug and fix this so that existing calls are not dropped? Thanks, Andrew
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