search for: 480s

...got rid of the firewall rule that opened the RTP ports. And then it dawned on me that I don't even need to open the 5060 port. The REGISTER requests established a UDP connection that the kernel's conntrack module was tracking anyway. The only issue was that the REGISTERs occurred only every 480s and the UDP connections were removed after 180s already. So at first I raised net.netfilter.nf_conntrack_udp_timeout_stream to 500. That worked. But I didn't really want to raise the default. So instead I added "qualify=yes" to the dtag_inbound peer. Now asterisk is sending an OPTION...

...s: 0x2 (CHECK CONDITION) Sense Bytes: 70 00 05 00 00 00 00 12 00 00 00 00 30 10 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 Sense Key: 0x5 Illegal Request, Segment 0 Sense Code: 0x30 Qual 0x10 (medium not formatted) Fru 0x0 Sense flags: Blk 0 (not valid) cmd finished after 0.010s timeout 480s cmd finished after 0.010s timeout 480s Fixating time: 0.013s cdrecord: fifo had 64 puts and 0 gets. cdrecord: fifo was 0 times empty and 0 times full, min fill was 100%. -- Tom Alsberg - hacker (being the best description fitting this space) Web page: http://www.cs.huji.ac.il/~alsbergt/ DI...

...terisk-dev@lists.digium.com>; asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> Subject: [Asterisk-Dev] adsi phone codes? Does anyone know off hand what codes should be used with the adsi phones in order to work with digium hardware and asterisk? The phones are the aastra 480s... _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev

...ists.digium.com>; asterisk-users@lists.digium.com <asterisk-users@lists.digium.com> >Subject: [Asterisk-Dev] adsi phone codes? > >Does anyone know off hand what codes should be used with the adsi phones in order to work with digium hardware and asterisk? The phones are the aastra 480s... >_______________________________________________ >Asterisk-Dev mailing list >Asterisk-Dev@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-dev > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.dig...

This is a really cool phone, except one problem, searched the archives and this was brought up before. Just wondering if anyone figured out how to solve it. I'm having the same problem as these previous posts... --------------- posted 06/09/03 Whenever I try using the voicemail through my ADSI display, it disables my # buttons. If I hit listen through the ADSI display, I can not delete

Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and

I do not want set allowguest=yes. The problem is, there is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. I have now the following addition to sip.conf. I think it is the only safe option. Or what would you say? [telekom](!) context=from-trunk type=peer defaultuser= authuser= remotesecret= fromdomain=tel.t-online.de