Two weeks ago 911 dispatch started having calling issues where they were getting ringing, then it would go to a busy signal.
On outgoing calls, the dispatcher establishes the SIP session within the enterprise asterix server, and it then flows to an Audiocodes media server that converts the SIP to Ethernet and sends it on to me, where I tandem the call to the PRI and out to the world.
I investigated it and found that it was regular and consistent, they would dial a call and if the party didn’t answer after 4 rings, the call would disconnect and dispatch would hear a busy signal. The telco translator looked at the examples I gave her, and in each circumstance the call was established and then she showed a normal termination. If the called party answers before four rings, the call continues.
The Asterix server guys tested some calls on their server yesterday and said they were receiving a disconnect command in their SIP session telling them to terminate the call.
I am thinking the problem almost has to lie in the Audiocodes box that lies between dispatch and my CS1000e, that there is some rogue timer within the SIP setup that is preventing more than four rings to be sent out before it is telling the caller to terminate the call.
On the other hand, can anyone think of a scenario where the CS1000e would cause the issue?
The folks that use the CS1000e are not experiencing this issue, only the dependent enterprise system, and the PRI that we tandem them out to the world across is the same as the local CS1000e users use.
Jim
"If I had known it would turn out like this, I would have become a locksmith" Albert Einstein
NCSS, NCTS, NCTE, CS1000E, Call Pilot
Avaya IP Office
Mitel 3300 Advanced, 5000, SX200, NuPoint, MiCollab, MBG
On outgoing calls, the dispatcher establishes the SIP session within the enterprise asterix server, and it then flows to an Audiocodes media server that converts the SIP to Ethernet and sends it on to me, where I tandem the call to the PRI and out to the world.
I investigated it and found that it was regular and consistent, they would dial a call and if the party didn’t answer after 4 rings, the call would disconnect and dispatch would hear a busy signal. The telco translator looked at the examples I gave her, and in each circumstance the call was established and then she showed a normal termination. If the called party answers before four rings, the call continues.
The Asterix server guys tested some calls on their server yesterday and said they were receiving a disconnect command in their SIP session telling them to terminate the call.
I am thinking the problem almost has to lie in the Audiocodes box that lies between dispatch and my CS1000e, that there is some rogue timer within the SIP setup that is preventing more than four rings to be sent out before it is telling the caller to terminate the call.
On the other hand, can anyone think of a scenario where the CS1000e would cause the issue?
The folks that use the CS1000e are not experiencing this issue, only the dependent enterprise system, and the PRI that we tandem them out to the world across is the same as the local CS1000e users use.
Jim
"If I had known it would turn out like this, I would have become a locksmith" Albert Einstein
NCSS, NCTS, NCTE, CS1000E, Call Pilot
Avaya IP Office
Mitel 3300 Advanced, 5000, SX200, NuPoint, MiCollab, MBG
