David P
2019-Dec-30 21:47 UTC
[asterisk-users] Handling a non-responsive peer after it answers
Response below... On Fri, Dec 27, 2019 at 12:02 PM David P <davidswalkabout at gmail.com> wrote:> > > > > I'm looking for a way of detecting in my dialplan when a peer becomes > > non-responsive after answering. [deleted] Is there a way to configure > > a handler for this state? > > > > We use v14.7.6 and we dial the peer this way: > > > > same => > > > n,Set(CHANNEL(hangup_handler_push)=${CONTEXT},handleHangupByCaller,1(args)) > > same => > > > n,Dial(${AddressToReachPeer},2,b(${CONTEXT}^afterDialingPeerLogIpOfCb^1(${UUID}^${StartEpoch}))) > > same => n,Goto(handle${DIALSTATUS},1) > > >"Joshua C. Colp" <jcolp at sangoma.com> replied:> [deleted] As for hanging up a call when the remote > goes away that depends on the channel driver. For SIP both chan_sip and > chan_pjsip provide session timers which use SIP messages to determine if > the call is no longer valid, or RTP timeout which hangs up the call if > media is not flowing for a period of time. These are configured in the > respective channel driver configuration file. >Thanks, Joshua. We want to check if a peer is responsive every few seconds, because it's a person-to-bot call and we want to respond gracefully if the bot fails. I tried adding rtptimeout=4 to the config of the peer in sip.conf, but this causes hangup during the person's turn. Then I looked into session timers, and found that https://issues.asterisk.org/jira/secure/attachment/28201/AsteriskSipSessionTimers.pdf says the shortest period supported for such checks is 90 seconds, which is much too long for us. Is there another option? Would it allow calling a script or playing a prompt on the way to hanging up? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20191230/b02288b4/attachment.html>
Joshua C. Colp
2019-Dec-30 22:05 UTC
[asterisk-users] Handling a non-responsive peer after it answers
On Mon, Dec 30, 2019 at 5:49 PM David P <davidswalkabout at gmail.com> wrote:> Response below... > > On Fri, Dec 27, 2019 at 12:02 PM David P <davidswalkabout at gmail.com> >> wrote: >> >> > >> > I'm looking for a way of detecting in my dialplan when a peer becomes >> > non-responsive after answering. [deleted] Is there a way to configure >> > a handler for this state? >> > >> > We use v14.7.6 and we dial the peer this way: >> > >> > same => >> > >> n,Set(CHANNEL(hangup_handler_push)=${CONTEXT},handleHangupByCaller,1(args)) >> > same => >> > >> n,Dial(${AddressToReachPeer},2,b(${CONTEXT}^afterDialingPeerLogIpOfCb^1(${UUID}^${StartEpoch}))) >> > same => n,Goto(handle${DIALSTATUS},1) >> > >> > > "Joshua C. Colp" <jcolp at sangoma.com> replied: > >> [deleted] As for hanging up a call when the remote >> goes away that depends on the channel driver. For SIP both chan_sip and >> chan_pjsip provide session timers which use SIP messages to determine if >> the call is no longer valid, or RTP timeout which hangs up the call if >> media is not flowing for a period of time. These are configured in the >> respective channel driver configuration file. >> > > Thanks, Joshua. > > We want to check if a peer is responsive every few seconds, because it's a > person-to-bot call and we want to respond gracefully if the bot fails. > > I tried adding > rtptimeout=4 > to the config of the peer in sip.conf, but this causes hangup during the > person's turn. > > Then I looked into session timers, and found that > https://issues.asterisk.org/jira/secure/attachment/28201/AsteriskSipSessionTimers.pdf > says the shortest period supported for such checks is 90 seconds, which is > much too long for us. > > Is there another option? Would it allow calling a script or playing a > prompt on the way to hanging up? >Those are the available options. There is no capability to call a script or play a prompt or anything like that that I can think of. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20191230/3647806d/attachment.html>
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