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Hello all, I have a linux machine with a SIP server (Asterisk) and 2 WAN interfaces (NATed) configured to do load balancing. I experienced problems with the SIP/RTP protocols and load balancing, because when initiating a call to an external SIP Host, a new RTP flow starts from the server to the Host, that sometimes uses another default route (due to the nexthop configuration). As i have two

Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with

Hi, Sorry to insist, but I would really like to be able to use Xen and my zaptel hardware all together in Dom0. I was just wondering if the 3.1 release could contain some changes compared to 3.0.4 related to IRQ handling or scheduling that could workout my problem. Thanks, François. ---- I actually first asked to asterisk mailing lists, and a few persons told me that it was Xen''s

Hi all, I would like to make a script/program that would be able to show lots of status information from my analog FXO lines (and FXS lines in the near future). Example of interesting status information: - Hook status: is there a call being made with that zap? - Voltage status: cable connected, voltage values (if possible), line ringing? - RX/TX Volume status I'm using a TDM400 card with

Hello all: I upgraded to 1.4.3 last night and use MySQL for CDR. I have noticed that 1.4.3 seems to log a lot of "crap" to CDR that 1.4.2 did not. I use a few macros in my dialplan to handle outgoing calls (lcr type stuff) and in addition to the proper CDR for the call itself I also have records to 's' in the same dest-context and entries to 's' in the default context.

Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar

Hello, i try to use iptables to mark packet and then to filter them with tc. Here is my script: iptables -t mangle -A PREROUTING -s 172.28.54.41/32 -p tcp -j MARK --set-mark 1 tc qdisc add dev eth0 handle ffff: ingress tc filter add dev eth0 parent ffff: protocol ip prio 1 handle 1 fw police rate 10000kbit burst 10000kbit mtu 1500k drop flowid :1 I can not use u32 because i have several

Hello! Is there any easy way to use the caller ID "display info" (CALLERID(name) in Asterisk) in dialplan just as we could use the number in: exten => _X./67803287, 1, <action> I have a SIP GSM device, and when a call comes in, it passes me the caller ID like so: -- Sip message Header: From: "67803287" <sip:gsm@192.168.10.1>;tag=... -- Asterisk variables:

Hello, I wanted to try some multipath setup with Julian''s patches, and when rebooting on the newly patched kernel, without any multipath routes (only one interface up and one default gateway), my SSH connections to the server freeze after a few seconds. I tried with and without bridges (I sometimes have a bridge between a tap device and my eth0) with the same results. When I close