Call does not hang up

[camilo.amb1234]

Good afternoon friends

I have the following inconvenience, when I call my sip trunk, the call goes through fine, I transfer that same call to another extension and the communication is ok, but when I hang up the call on the IP phone, the call does not hang up on the device from where I made it. the call, for example my cell phone, that call remains there as if it were still active.

[Pooh]

Good afternoon friends

Good evening.

I have the following inconvenience, when I call my sip trunk, the call goes
through fine,

Good start.

I transfer that same call to another extension

How?

and the communication is ok, but when I hang up the call on the IP phone,
the call does not hang up on the device from where I made it.

So, this is only after a transfer? Does a simple “inbound call - answer -
hang up” work as expected?

the call, for example my cell phone, that call remains there as if it were
still active.

How disappointing. There is probably something wrong with your dialplan.

Give us enough information to understand how your calls are being routed and
transferred and we might be able to suggest a solution to your problem.

As it is, though, we have no idea how you have set your system up and
therefore we have no idea what might be wrong with it.

Antony.

[david551]

Which phone is THE IP phone?

There are many ways of doing transfers when a SIP endpoint is the transferor:

• Feature codes.
• SIP REFER (for answered call blind transfers)
• SIP INVITE, followed by REFER/Replaces (answered call attended transfers, and also used, in an automated way, for unattended ones)
• Status 302, for unanswered (always blind) transfers

You didn’t say how you called your trunk (actually trunks don’t exist in SIP, but I assume you mean via an ITSP).

Do the extensions dial SIP devices?

In a pure SIP environment, the likely reason for hangup not to work is that the wrong Contact address was sent, but you would either expect the same problem if the call has hungup before the transfer, at least if the call was made directly to the final endpoint.

If analogue devices are involved, there are other mechanisms.

We basically need much more information about your configuration to do more than guess. We may well need SIP protocol traces.

[sammiesoft]

I am having a similar problem. Although I am using Node ARI. This is my code.

let bridge = await channel._client.Bridge().create();
await bridge.addChannel({channel: channel.id});

let outboundChannel = await channel._client.Channel().create({
    endpoint : `PJSIP/dsl-ivr`,
    app : 'outbound',
});

await bridge.addChannel({channel: outboundChannel.id});

await outboundChannel.dial({timeout: timeout});

const channelChannelHangupRequestHandler = (event: Event) => {
    outboundChannel.removeListener('ChannelHangupRequest', 

outboundChannelChannelHangupRequestHandler);
outboundChannel.hangup();
};

const outboundChannelChannelHangupRequestHandler = (event: Event) => {
    channel.removeListener('ChannelHangupRequest', channelChannelHangupRequestHandler);
    channel.hangup();
};

channel.once('ChannelHangupRequest', channelChannelHangupRequestHandler);

outboundChannel.once(‘ChannelHangupRequest’,outboundChannelChannelHangupRequestHandler);