[Asterisk-Users] dropped calls

[Paulo Loureiro]

Well, after days chasing this ghost, the problem seems to be solved:

- There was a problem with the NTP server (not responding to clients due
to config restrictions)
- upgrade gs firmware to latest available: 1.4.50.

If the NTP server is down gs phones keep trying to reach it and drop
some of the packets received from * during a call. * thinks the phone is
"dead" and drops the call. 

Anyone can confirm this?

--- Paulo Loureiro.



On Fri, 2004-03-05 at 19:24, Bartosz Jozwiak wrote:
> 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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> >
> 
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-- 
Cumprimentos,

--- Paulo Loureiro
Netmania