Savoy, Kevin - Williston, ND
2007-Jan-16 10:08 UTC
[asterisk-users] How to detect long calls
Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 2151 bytes Desc: image001.gif Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070116/069b24b6/attachment.gif
Savoy, Kevin - Williston, ND wrote:> We have been running an Asterisk box with 1.2.9.1 on it since August in > a call center environment. We use the Asterisk box as an IVR and then > pass the calls on to a Nortel Option 11C. Today we found in our long > distance bill two calls that lasted a VERY long time. One was 58 hours > and another was 38 DAYS!!! > > > > Nortel does not show this call being that long. Obviously the person > that called in didn?t hold the line for 58 days so somehow between > Asterisk and MCI the call got stuck open and didn?t hang up on the network. > > > > My question is two parts, part one, has anyone heard of anything like > this where a call doesn?t hang up properly and seems ?stuck? in the > system. Part two is there anyway to monitor in Asterisk the length of > all active calls and then if a call lasts longer then, say one hour, we > could send off a text message or warning. > >Hi , similiar thing happend to me. Try looking at the L() optin in Dial. I define a max call time, say few hours, then warn every x seconds, then cut the call. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.
>>>>> "KS" == "Savoy, Kevin <- Williston, ND" <ksavoy@novo1.com>> writes:KS> We have been running an Asterisk box with 1.2.9.1 on it since KS> August in a call center environment. We use the Asterisk box as an KS> IVR and then pass the calls on to a Nortel Option 11C. Today we KS> found in our long distance bill two calls that lasted a VERY long KS> time. One was 58 hours and another was 38 DAYS!!! There has been some excellent suggestions in this thread. I just want to add one. Sometimes a SIP packet can get lost or a phone rebooted without closing a call properly. Then the call will just stay open forever. You can solve that with rtptimeout=3600 or something similar in sip.conf. Obviously it only works when the rtp stream is actually going through Asterisk, and it will also kill the call if a Snom phone turns the microphone off for an hour (Snom phones do silence suppression unconditionally when you press the mute button). Still, even with those limitations it works nicely for us. At least as a stopgap until Asterisk gets TCP-support for SIP. /Benny
Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 2151 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070116/c1aa1d25/attachment.gif
Skipped content of type multipart/related-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: Esta =?ISO-8859-1?Q?=E9?= uma parte de mensagem assinada digitalmente Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070116/6c512581/attachment.pgp
----- Original Message ----- From: "Benny Amorsen" <benny+usenet@amorsen.dk> To: <asterisk-users@lists.digium.com> Sent: Wednesday, January 17, 2007 12:10 AM Subject: [asterisk-users] Re: How to detect long calls>>>>>> "KS" == "Savoy, Kevin <- Williston, ND" <ksavoy@novo1.com>> writes: > > KS> We have been running an Asterisk box with 1.2.9.1 on it since > KS> August in a call center environment. We use the Asterisk box as an > KS> IVR and then pass the calls on to a Nortel Option 11C. Today we > KS> found in our long distance bill two calls that lasted a VERY long > KS> time. One was 58 hours and another was 38 DAYS!!! > > There has been some excellent suggestions in this thread. I just want > to add one. Sometimes a SIP packet can get lost or a phone rebooted > without closing a call properly. Then the call will just stay open > forever. You can solve that with rtptimeout=3600 or something similar > in sip.conf. Obviously it only works when the rtp stream is actually > going through Asterisk, and it will also kill the call if a Snom phone > turns the microphone off for an hour (Snom phones do silence > suppression unconditionally when you press the mute button). > > Still, even with those limitations it works nicely for us. At least as > a stopgap until Asterisk gets TCP-support for SIP. > > > /Benny > >I know this post s old but I will add my 2c. In sip.conf you can use silencedetecthangup=x. Have a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk%20sip%20silencedetecthangup