[Freeswitch-users] ISDN to SIP via Freeswitch - 408 Timeout and Terrible line noise

samwise samwise+fsu at bagshot-row.org
Fri May 21 09:03:22 PDT 2010


Hi,

I've just joined a team which had an Asterisk 1.6.0.1 desktop machine
with a Digium TE122 card running CentOS 4.7 to bridge calls from, as
far as I can tell, an E1 PRI line (with 10 DDI numbers) and software
SIP UAs.  Unfortunately, the machine died recently and will no longer
boot from hard disk.

So I've taken the card out and placed it in a HP Compaq ProLiant ML350
G3 Server with 1 GB RAM running CentOS 5.4 (Final).  I've set it up to
run the Freeswitch 1.0.6 release with the OpenZAP module enabled and
pre-requisites dahdi-linux-complete-2.3.0+2.3.0 and libpri-1.4.10.2.
All these components were built from source packages.

I installed the default configuration files and configured everything
as far as I could by following the instructions on the Freeswitch
wiki.  I've managed to get as far as successfully being able to dial
one of the DDI numbers and getting the call to be delivered to an
X-Lite SIP UA I have registered with Freeswitch.

Unfortunately, I have two issues:

1) I have configured a SIP user account (Greg) but when I try to
register X-Lite, it consistently returns a Registration error: 408 -
Request Timeout.  However, if I leave X-Lite alone, most of the time
it will eventually (after quite some time) re-register with the server
of it's own accord, allowing me to try a test call from the PSTN.

2) Whilst the SIP UA can then hear the call fine, there is a *lot* of
noise on the PSTN end of the call - so much so that it's basically
unusable, even though I can hear some things from the SIP end.

Some info on the kernel/CPU included belom and some possibly useful
logs / config files available here:

openzap.conf: http://pastebin.freeswitch.org/13019
openzap.conf.xml: http://pastebin.freeswitch.org/13020
Greg.xml: http://pastebin.freeswitch.org/13021
61522x.xml: http://pastebin.freeswitch.org/13022
freeswitch.log: http://pastebin.freeswitch.org/13023

My question is where to start looking to diagnose these problems?  I
can provide whatever logs may be appropriate though I can't see any
major problems in the freeswitch.log currently.

With regards to the line noise issue, I read in the OpenZAP FAQ that
unlike Asterisk, Freeswitch does not include a software echo
canceller.  Could that be related to the line noise?  Should I attempt
to install OSLEC to improve the quality?

Thanks for any help - I'm fairly new to the world of telephony
tecnologies like ISDN,

Sam.

# uname -a
Linux bob 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686
i386 GNU/Linux
# cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 8
model name      : Pentium III (Coppermine)
stepping        : 6
cpu MHz         : 864.011
cache size      : 256 KB
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 2
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca
cmov pat pse36 mmx fxsr sse up
bogomips        : 1728.02



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