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" <[EMAIL PROTECTED]>
To: <asterisk-users@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) [EMAIL PROTECTED] 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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