Hi,
My configuration: Sipura 2000
Debian/Sarge Asterisk 1.0.1 built by msp@toshiba on a i686 running Linux
I am calling 612@fwd.pulver.com which is Daytimephoneline of pulver.com and for
the first second the connection seems to be ok and I hear: <thu .. rsda
..> and nothing more, which suppose to mean <Thursday, ...>. The echo
line from 613@fwd.pulver.com does work the same. I never got an echo back, the
connection seems to stay online.
When I call somebody on pulver.com than it works. When I do a count <one,
two, three, four, five, six,...> the person receives: <one, two, ..., ...,
fi..e,...ten> . Same other way around. The connection will still be alive.
Here my * output with (asterisk -vvvvvvvgc):
-- Executing SetCallerID("SIP/2201-f8d6", "password") in new
stack
-- Executing SetCIDName("SIP/2201-f8d6",
""username"") in new stack
-- Executing Dial("SIP/2201-f8d6", "SIP/612@fwd.pulver.com")
in new stack
-- Called 612@fwd.pulver.com
-- SIP/fwd.pulver.com-29d8 is ringing
-- SIP/fwd.pulver.com-29d8 answered SIP/2201-f8d6
-- Attempting native bridge of SIP/2201-f8d6 and SIP/fwd.pulver.com-29d8
== Spawn extension (home, 7612, 3) exited non-zero on 'SIP/2201-f8d6'
my sip file starts of with:
[general]
bindaddr = 192.168.2.200
port = 5060 ; Port to bind to
disallow= all ; Disallow all codecs
allow = gsm
allow = ilbc
;allow = ima-adpcm
allow = ulaw
allow = alaw
externip = ipthatproviderhasgiventome
localnet = 192.168.2.0/255.255.255.255
context = from-sip
Is there any differense between 1st call and 2nd call? Could that be a bad QoS
Problem? The Internetconnection is 256k/s and there is actually not much traffic
on.
Robert.
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