[Asterisk-Users] dropped calls

[Bartosz Jozwiak]

I have couple of GS phone and CISCO 7960.
The funny thing is that two of that GS phone keep disconnecting and also
CISCO 7960 phone keeps disconnecting.
But the problem appear month ago! This is really strange!

Bart



> Hello,
>
> I'll try that, but why on earth gs phones with the same firmware on
> another * server, work with no problem?
>
> I've failed to state I'm using zaprtc, since there is no digium hardware
> on the server. Does it matter?
>
> Thanks,
>
> --- Paulo.
>
>
> On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote:
> > There is new firmware that may help
http://www.grandstream.com/BETATEST/.
> > Grandstream acknowledges this problem. They say it is a codec issue with
> > asterisk. I don't know if this update addresses this problem but it may
be
> > worth a try.
> >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com
> > > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> > > Paulo Loureiro
> > > Sent: Friday, March 05, 2004 10:26 AM
> > > To: asterisk-users at lists.digium.com
> > > Subject: [Asterisk-Users] dropped calls
> > >
> > > Hello list,
> > >
> > > I'm getting droped calls on an asterisk installation. When on GS phone
> > > dials another one, the call is dropped after some (usually
> > > random) time
> > > but most of the tome within 3 to 20 seconds.
> > > I think the cause is stated on the logs, see bellow, and is
> > > related with
> > > the message  "Didn't get a frame from channel: SIP/3805-df43", but I
> > > can't figure why.
> > >
> > >
> > > asterisk logs:
> > > -------------------------------------
> > > Mar  5 15:57:26 DEBUG[1116957488]: build_route: Contact hop:
> > > <sip:192.168.60.106>
> > > Mar  5 15:57:26 VERBOSE[1217669936]:     -- SIP/3805-df43 answered
> > > SIP/-08122450
> > > Mar  5 15:57:26 VERBOSE[1217669936]:     -- Attempting native
> > > bridge of
> > > SIP/-08122450 and SIP/3805-df43
> > > Mar  5 15:57:26 DEBUG[1116957488]: Stopping retransmission on
> > > 'c9f88915cb5c25fd at 192.168.60.107' of Response\ 25663: Found
> > > Mar  5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524
> > > Mar  5 15:57:26 DEBUG[1217669936]: Ooh, format changed from
> > > UNKN to ULAW
> > > Mar  5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel:
> > > SIP/3805-df43
> > > Mar  5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels
> > > SIP/-08122450 and SIP/3805-df43
> > > Mar  5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse
> > > counter
> > > Mar  5 15:57:38 DEBUG[1217669936]:  is not a local user
> > > Mar  5 15:57:38 VERBOSE[1217669936]:   == Spawn extension
> > > (local, 3805,
> > > 1) exited non-zero on 'SIP/-0812245\0'
> > > -----------------
> > >
> > > The scenario:
> > > 1 server (redhat 9),  asterisk (stable) and a 2 x hisax ISDN BRI.
> > > One of the BRI boards is used to dial out (ppp) on one channel and a
> > > mgetty on the other channel. The other board is in ptp and used by *.
> > > The phones are Grandstream BT101 and Handytone and are all on
> > > a switched
> > > network (3 procurve switches, stacked).
> > >
> > > The configs are ok, since the same files on another server work ok (no
> > > dropped calls), but I can post them if needed.
> > >
> > >
> > > Any help will be greatly appreciated.
> > >
> > > Thanks in advance,
> > >
> > >
> > >
> > > --- Paulo Loureiro
> > >
> > >
> > >
> >
> > _______________________________________________
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> -- 
> Cumprimentos,
>
> --- Paulo Loureiro
> Netmania
>
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