I have ran into a case using 1.4.22 where a SIP call to an asterisk
client (running a slow PC) to ALSA
does not hangup the call when it is done. The server is using call files
to initiate the call, the client answers on
the ALSA port, the server plays the message and hangs up.
I found that SOMETIMES -its hard to recreate - that the slow pc keeps
the SIP channel active. further calls in
are getting a busy signal and the one call is NEVER hung up.
How can I detect this and hang up the channel on the slow PC. I verified
on the server that it thinks NO calls are active.
my context looks like this:
[mycontext]
exten => s,1,ChanIsAvail(Console/Dsp)
exten => s,n,GotoIf($["${AVAILCHAN}" =
""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Hangup
[smvoice-busy]
exten => s,1,playtones(busy)
exten => s,1,wait(10)
exten => s,1,Hangup
Thanks,
Jerry
hi, try to set the rtptimeout value in sip.conf to a resonable value - so asterisk will kill the channels if it does not receive rtp traffic for the specified time regards, Wolfgang Jerry Geis schrieb:> I have ran into a case using 1.4.22 where a SIP call to an asterisk > client (running a slow PC) to ALSA > does not hangup the call when it is done. The server is using call files > to initiate the call, the client answers on > the ALSA port, the server plays the message and hangs up. > > I found that SOMETIMES -its hard to recreate - that the slow pc keeps > the SIP channel active. further calls in > are getting a busy signal and the one call is NEVER hung up. > > How can I detect this and hang up the channel on the slow PC. I verified > on the server that it thinks NO calls are active. > > my context looks like this: > [mycontext] > exten => s,1,ChanIsAvail(Console/Dsp) > exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1) > exten => s,n,Playback(beep) > exten => s,n,Dial(Console/dsp) > exten => s,n,Hangup > > [smvoice-busy] > exten => s,1,playtones(busy) > exten => s,1,wait(10) > exten => s,1,Hangup > > Thanks, > > Jerry > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> > hi, > > try to set the rtptimeout value in sip.conf to a resonable value - so > asterisk will kill the channels if it does not receive rtp traffic for > the specified time > > regards, > Wolfgang >I uncommeted the rtptimeout=60 value in sip.conf and did a reload. It still hasnt dropped the dead call after a couple minutes now... Do I have to stop and start again? Was hoping it would just drop the call and continue on. Jerry
Jerry Geis wrote:>> hi, >> >> try to set the rtptimeout value in sip.conf to a resonable value - so >> asterisk will kill the channels if it does not receive rtp traffic for >> the specified time >> >> regards, >> Wolfgang >> >> > I uncommeted the rtptimeout=60 value in sip.conf and did a reload. > It still hasnt dropped the dead call after a couple minutes now... > > Do I have to stop and start again? Was hoping it would just drop the > call and continue on. > > JerrySounds like the problem is that the slow computer is no longer accepting calls after the first. Is Asterisk running on that machine as well? If so, check to see what it says about the sip channels. If not, you will need to look into the software running on that machine and try to figure out why it is either not hanging up or why it is dieing after it gets a call. -Brent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090120/fc2c9bc0/attachment.htm