[Asterisk-Users] Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help

Serge Schumacher serge at vonet.lu
Fri Dec 31 06:08:29 MST 2004


Might be related to the musiconhold files  using different encoding rates ?

Just an idea, also a newbie :)

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paid Up
Sent: vendredi 31 décembre 2004 14:01
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Host IP address, Crash on startup, Console grabs
soundcard - Newbie needs help

I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1
(see config below) and with a bit of 
messing about using sample config, have been able to make the test call to
device 1000, and also through to the IAX 
test number at Digium. However, operation is extremely flaky - I cannot
reliably startup and use the system on a 
regular basis. I have several problems listed below and would appreciate any
insights the experts can offer.   
   
Problem 1)   
The server is given its IP address using DHCP from my residential DSL
gateway. The DNS settings are those from my   
ISP. Therefore the name allocated to the server (Vigor15) cannot be resolved
to an IP address using DNS lookup.   
Other programs do not seem to be affected by this. I've fixed this by adding
the name into the /etc/host.conf file,   
but wondered if this was an issue with the application (asterisk) or more
generally my setup. I'm not sure if this   
is related to a problem where SIP, IAX protocols are set to listen on IP
address 0.0.0.0 as in 2 below.   
   
   
Problem 2)   
SIP softPhones can't register. I think this may be due to listening on the
wrong IP address 0.0.0.0:5060. Here's the   
log during startup:   
   
chan_sip.so] => (Session Initiation Protocol (SIP))   
  == Parsing '/etc/asterisk/sip.conf': Found   
  == SIP Listening on 0.0.0.0:5060   
  == Using TOS bits 0   
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))   
  == Registered application 'SIPDtmfMode'   
  == Registered application 'SIPAddHeader'   
  == Registered application 'SIPGetHeader'   
   
Problem 3)   
Sometimes the program crashes during (at end of) the startup sequence.
Warning about "flexibel rate not heavily   
tested". Is this just a codec I can configure off/disable, or is this a
crucial part of the system that will   
hopefully be fixed soon. I got this problem both with the latest stable
release 1.0.1-2 (included in Mandrake) as   
well as the latest CVS-HEAD version checked out and rebuilt. The crash might
be related to (4) below.   
   
   
cdr_manager.so] => (Asterisk Call Manager CDR Backend)   
  == Parsing '/etc/asterisk/cdr_manager.conf': Found   
  == Parsing '/etc/asterisk/enum.conf': Found   
Asterisk Ready.   
*CLI> Ouch ... error while writing audio data: : Broken pipe   
Segmentation fault (core dumped)   
[root at Vigor15 david]# Warning, flexibel rate not heavily tested!   
   
Problem 4)   
Asterisk grabs the sound card for console use by default on startup. Its
therefore not possible/easy to run KPhone   
or similar which also requires that resource. How can I turn off/stop
asterisk trying to use the soundcard, and what   
are the implications.   
   
TIA   
Paidup   
   
System is P3 800MHz with 512MB ram, 19GB Disk, 100MBit Ethernet. Mandrake
Linux Official 10.1. Similar problems with   
both recent stable release asterisk 1.0.1-2 and CVS_HEAD. SOftphone is
KPhone (using SIP) on same machine.     
      
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