Cant able to have IVR Working using third party Gateway

Asterisk Asterisk SIP
1

I have sip trunk between freepbxNow and Cisco call manager, internal calls between them is working fine, the issue is when I call the IVR from outside PSTN to (FXO) line h323 gateway, IVR is working but when I disconnected asterisk keeps the line used, i’m getting busy tone if I tried to call the number again, anyone have solution for this issue?

2

I’m a bit confused about the configuration, in particular why you are asking about Asterisk when the issue seems to be with a third party gateway, and who is clearing the IVR call.

However, it is difficult to detect when an incoming analogue line call is cleared by the remote side, as) the caller may not send any disconnect supervision signal, or the device terminating the analogue line may not have been configured to rcognize the provider’s signal. Typical disconnect supervision signals are polarity reversals, and temporary removals of battery voltage, but if Asterisk itself terminates the line, it can also listen for busy tone.

For an incoming call that is terminated by the callee, traditionally the network will not send any disconnect supervision to the caller until the callee has been on hook for about three minutes, to allow them to put down one phone and pick up another, although, in the UK, that has been reduced to a few seconds in many places.

Generally, to get reliable disconnect supervision to and from the public network, you need to interface with ISDN. SIP provides will always do this. Even then, a callee hangup will initially be signalled as a CLEAR, but that can be cancelled by a REANSWER, so some device treat these as HOLD and UNHOLD, and wait for the network RELEASE before hanging up the call.

1 Like
4

Thanks for your reply,

if I call to IVR from internal ext (CUCM Side) and disconnect the call, the ivr is hangup just fine, please see the log blew from asterisk

– Executing [h@ivr-1:1] Hangup(“PJSIP/sip-to-cucm-00000019”, “”) in new sta ck
== Spawn extension (ivr-1, h, 1) exited non-zero on ‘PJSIP/sip-to-cucm-0000001 9’

this execution is not coming in above scenario

5

SIP always supports disconnect supervision. Your issue will be that the FXO gateway is unable to recognize disconnect supervision from the PSTN, or the PSTN is not providing any.