[Asterisk-Users] dropped calls

[Paulo Loureiro]

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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-- 
Cumprimentos,

--- Paulo Loureiro
Netmania