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