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

[Claus Futtrup]

Hi,

1) 0.0.0.0 just means listning on all interfaces and their ip adresses, not 
a problem.
2) Do a set verbose 100 to see if you have any communication with the sip 
phones or startup asterisk with asterisk -vvvddddggg
3) This is because a MPG3 file used for music on hold isn't support or that 
the Mandrake mpg123 is a wrong version
4) Try unloading the ALSA module in modules.conf

Kind Regards

Claus

----- Original Message ----- 
From: "Paid Up" <paidup at post.com>
To: <asterisk-users at lists.digium.com>
Sent: Friday, December 31, 2004 2:00 PM
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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