We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian
-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Thursday, July 08, 2010 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not detecting hangup We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian -- -- First thought is that you can put a timeout on your calls, but that is just a "band-aid". _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Julian > Lyndon-Smith > > We have had 20 calls over the last month where the SIP channel has not > identified that the person on the receiving end has hung up. > > Is there a way of fixing this ?On Thu, 8 Jul 2010, Danny Nicholas wrote:> First thought is that you can put a timeout on your calls, but that is > just a "band-aid".Also not fixing the source of the problem, but rtpholdtimeout and rtptimeout may help. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
That looks like the option that will help a lot. Thanks. On 8 July 2010 23:21, Steve Edwards <asterisk.org at sedwards.com> wrote:>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Julian >> Lyndon-Smith >> >> We have had 20 calls over the last month where the SIP channel has not >> identified that the person on the receiving end has hung up. >> >> Is there a way of fixing this ? > > On Thu, 8 Jul 2010, Danny Nicholas wrote: > >> First thought is that you can put a timeout on your calls, but that is >> just a "band-aid". > > Also not fixing the source of the problem, but rtpholdtimeout and > rtptimeout may help. > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards ? ? ? sedwards at sedwards.com ? ? ?Voice: +1-760-468-3867 PST > Newline ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?Fax: +1-760-731-3000 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > ? ? ? ? ? ? ? http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >