[Asterisk-Users] Mixing meetme conferences
Matt Florell
astmattf at gmail.com
Mon Jun 5 13:48:15 MST 2006
What you describe is not a problem with meetme, it is an issue with
the old call not hanging up from the manager API command that VICIDIAL
sends. This is usually caused by a misconfiguration or the system
being overloaded.
MATT---
On 6/5/06, Erick Perez <eaperezh at gmail.com> wrote:
> The call mix occurs randomly when in VICIDIAL an agent get a call,
> then dispositions/hangup the call (example: phone number is an
> aswering machine) and when the agent returns to the main menu and
> instantly gets another call, the call mixes the new one with the old
> one.
>
>
> Here is the output
>
> centos*CLI> show channels concise
> SIP/209.120.202.94:5060-34f4!default!!1!Down!AppDial!(Outgoing
> Line)!0000000000!!3
> !!(None)
> Local/917342077795 at default-5065,2!default!917342077795!2!Ring!Dial!SIP/17342077795
>
> @209.120.202.94:5060|55|o!0000000000!!3!0!(None)
> Local/917342077795 at default-5065,1!default!s!1!Down!(None)!!0000000000!!3!!(None)
> SIP/209.120.202.94:5060-f3e5!default!!1!Down!AppDial!(Outgoing
> Line)!0000000000!!3
> !!(None)
> Local/917342160725 at default-f57a,2!default!917342160725!2!Ring!Dial!SIP/17342160725
>
> @209.120.202.94:5060|55|o!0000000000!!3!0!(None)
> Local/917342160725 at default-f57a,1!default!s!1!Down!(None)!!0000000000!!3!!(None)
> SIP/209.120.202.94:5060-8cd4!default!!1!Down!AppDial!(Outgoing
> Line)!0000000000!!3
> !!(None)
> Local/917329965709 at default-cfea,2!default!917329965709!2!Ring!Dial!SIP/17329965709
>
> @209.120.202.94:5060|55|o!0000000000!!3!13!(None)
> Local/917329965709 at default-cfea,1!default!s!1!Down!(None)!!0000000000!!3!!(None)
> SIP/209.120.202.94:5060-de9e!default!!1!Down!AppDial!(Outgoing
> Line)!0000000000!!3
> !!(None)
> Local/917329964777 at default-0134,2!default!917329964777!2!Ring!Dial!SIP/17329964777
>
> @209.120.202.94:5060|55|o!0000000000!!3!23!(None)
> Local/917329964777 at default-0134,1!default!s!1!Down!(None)!!0000000000!!3!!(None)
> SIP/209.120.202.94:5060-0316!default!8600053!1!Up!MeetMe!8600053!!!3!16!(None)
> SIP/209.120.202.94:5060-2041!default!!1!Up!Bridged
> Call!SIP/1003-1bb7!1003!!3!!SIP
> /1003-1bb7
> SIP/1003-1bb7!default!445712152808714!2!Up!Dial!SIP/12152808714 at 209.120.202.94:506
>
> 0|55|o!1003!!3!531!SIP/209.120.202.94:5060-2041
> Zap/pseudo-1102943127!unused!s!1!Rsrvd!(None)!!!!3!!(None)
> SIP/1015-6c25!default!8600053!1!Up!MeetMe!8600053!!!3!1476!(None)
> Zap/pseudo-224692978!unused!s!1!Rsrvd!(None)!!!!3!!(None)
> SIP/1016-0fc2!default!8600054!1!Up!MeetMe!8600054!!!3!5066!(None)
> Zap/pseudo-1800952753!unused!s!1!Rsrvd!(None)!!!!3!!(None)
> SIP/1012-6437!default!8600057!1!Up!MeetMe!8600057!!!3!6324!(None)
> Zap/pseudo-531907954!unused!s!1!Rsrvd!(None)!!!!3!!(None)
> SIP/1014-1a13!default!8600059!1!Up!MeetMe!8600059!!!3!6440!(None)
> Zap/pseudo-1285637729!unused!s!1!Rsrvd!(None)!!!!3!!(None)
> SIP/1019-929e!default!8600051!1!Up!MeetMe!8600051!!!3!12895!(None)
>
>
> On 6/5/06, Matt Florell <astmattf at gmail.com> wrote:
> > It would help if you included some more information, maybe like the
> > output from "show channels concise" from Asterisk and then a summary
> > of which channels and/or meetmes are mixing audio.
> >
> > I have not run into any issues with audio from one meetme bleeding
> > into another, but since you mention meetme and call center I assume
> > you are using VICIDIAL which might mean you are having some issues
> > with your agent's not logging out properly.
> >
> > MATT---
> >
> > On 6/5/06, Erick Perez <eaperezh at gmail.com> wrote:
> > > Have anyone experienced mixed meetme conferences?
> > > Im running a 12 seat call center outbound only. Asterisk 1.2.8,
> > > SIP/ulaw at the phones, SIP/ulaw to the SIP terminator.
> > >
> > > Machine: pentium Dual core 2.0ghz (533 fsb) , 1 GB RAM (533mhz ddr) ,
> > > 2 SATA 80GB DISK in RAID 0 via software RAID driver. Two Intel PRO
> > > 10/100 NIC, IRQs are separated, an X100P so not to use ztdummy.
> > > Motherboard is an Intel 945GNT.
> > >
> > >
> > > output of vmstat
> > > procs -----------memory---------- ---swap-- -----io---- --system-- ----cpu----
> > > r b swpd free buff cache si so bi bo in cs us sy id wa
> > > 2 0 0 493284 44996 290924 0 0 2 16 247 116 3 2 95 0
> > >
> > >
> > > cat /proc/interrupts
> > >
> > > CPU0 CPU1
> > > 0: 24778955 24730655 IO-APIC-edge timer
> > > 1: 8 0 IO-APIC-edge i8042
> > > 8: 76 90 IO-APIC-edge rtc
> > > 9: 0 0 IO-APIC-level acpi
> > > 12: 66 0 IO-APIC-edge i8042
> > > 14: 88370 86773 IO-APIC-edge ide0
> > > 15: 74129 72701 IO-APIC-edge ide1
> > > 169: 0 0 IO-APIC-level uhci_hcd
> > > 185: 11 7692208 IO-APIC-level eth1, uhci_hcd
> > > 193: 0 0 IO-APIC-level uhci_hcd
> > > 201: 0 0 IO-APIC-level uhci_hcd
> > > 209: 24772638 24706144 IO-APIC-level wcfxo
> > > 217: 4126715 0 IO-APIC-level eth0
> > > NMI: 49705668 49705619
> > > LOC: 49515422 49526061
> > > ERR: 0
> > > MIS: 0
> > >
> > >
> > > --
> > > ------------------------------------------------------------
> > > Erick Perez
> > > Panama Sistemas
> > > Integradores de Telefonia IP y Soluciones para centros de datos
> > > Panama, Republica de Panama
> > > ------------------------------------------------------------
> > > _______________________________________________
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>
>
> --
> ------------------------------------------------------------
> Erick Perez
> Panama Sistemas
> Integradores de Telefonia IP y Soluciones para centros de datos
> Panama, Republica de Panama
> ------------------------------------------------------------
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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