[asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

Jonathan H lardconcepts at gmail.com
Wed May 26 13:29:39 CDT 2021

I think I can confidently say, after most of a day and reading the following....


... that Asterisk doesn't like mixing autoHangup with AGI, and there
appears to be no way of the ts-agi library I'm using knowing that it
has autoHungup, so it can't close the AGI connection which seems to
release Asterisk to hangup properly.

I had thought that the AGIEXITONHANGUP variable might help, but it
appears to do nothing, although I'm unsure if I'm setting it correctly

Here it says the flag is "1"

And here it says the flag is "yes"

That said, I wish I could use ARI not AGI but without the current
media offset available on ARI, I need AGI :)

So for now, the workaround is to forget about setting autohangup, and
just hangup the caller manually at the point they don't over-ride the


On Wed, 26 May 2021 at 18:01, Joshua C. Colp <jcolp at sangoma.com> wrote:
> On Wed, May 26, 2021 at 1:58 PM Jonathan H <lardconcepts at gmail.com> wrote:
>> I have also tried configuring pjsip wizard like this.
>> endpoint/rtp_timeout=5
>> And I see this shortly after the "hangup" command has been sent, so
>> that part is working:
>> [May 26 17:36:37] NOTICE[1276]: res_pjsip_sdp_rtp.c:150
>> rtp_check_timeout: Disconnecting channel
>> 'PJSIP/fromvoipfone-206-0000000b' for lack of audio RTP activity in 5
>> seconds
>> But, again, it doesn't disconnect. The line stays open. And yes, my
>> fallthrough after agi is
>> same => n, Hangup()
>> Also, apparently I now have a load of channels, which won't even hangup with
>> channel request hangup all
>> Requested Hangup on channel 'PJSIP/fromvoipfone-206-0000000b'
>> Requested Hangup on channel 'PJSIP/fromvoipfone-206-0000000a'
>> Requested Hangup on channel 'PJSIP/fromvoipfone-206-00000009'
>> ...and wait.. and then...
>> Channel              Location             State   Application(Data)
>> PJSIP/fromvoipfone-2 s at test:2             Up      AGI(agi://localhost:3456)
>> PJSIP/fromvoipfone-2 s at test:2             Up      AGI(agi://localhost:3456)
>> PJSIP/fromvoipfone-2 s at test:2             Up      AGI(agi://localhost:3456)
>> 3 active channels
>> 3 active calls
>> So they just won't die.
>> Asterisk 18.4.0 - worth filing a bug?
> Is your AGI closing the connection or are you expecting Asterisk to drop it? (I'm not that familiar with FastAGI or AGI these days, just wondering what happens if you drop the connection)
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

More information about the asterisk-users mailing list