Call Hangup with One Channel in Asterisk ARI

Hi,

I am using Asterisk 15.5 and python ARI, I have made a custom voicemail application in ARI. When an incoming call comes, I answer the incoming call and record the channel in ARI.

I am facing an issue that after 32 seconds call is being disconnected every time. Asterisk is sending bye response to SIP Trunk provider.

I have also tried to play media at a certain interval but then also the call is getting disconnected.

Are you behind NAT? Have you set up the respective SIP channel driver to know it is behind NAT?

Yes JCOLP,

But my incoming and outgoing calls are working fine.

32 seconds is the characteristic time for failing to confirm the sending of a packet, which is usually due to firewall or NAT configuration problems.

You need to get a protocol logging for your channel technology driver (which you haven’t identified), although the basic reason for the hangup will normally be reported with default logging settings.

I am using PJSIP as channel technology

Hi Jcolp and David,

My incoming calls are also getting disconnected around 32 seconds. What could be the issue?

Do I need to check firewall or Asterisk config?

You would need to check both, starting with the Asterisk configuration.