similar to: Threading troubles 1.4.5 & IAX2-> SIP (FreeBSD specific??)

Hi there, I'm in a situation where some of my clients have all these obscurity things with ssh, like putting it on a different port, or a different user to login for this specific host not mention several don't have proper DNS names and and and. Life is to short to debate it with the client so rather get in line, use ssh_config, and continue to serve the client. However, now I have the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 No, even if bandwidth is your concern I would say that --checksum is wrong. Maybe if bandwidth is so scarse that a few KB vs a few MB equates to dollars then sure, use --checksum. Otherwise, letting rsync re-delta-xfer everything is certainly faster and not much more bandwidth intensive than --checksum. Plus that is only if you screwed up and ran

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Normally, I would say that --checksum is actually slower than just letting rsync re-copy everything and therefore is almost always the wrong thing to do. However, in this case, you really don't want to overwrite the running OS even with files that are essentially the same. So, if the system is running from that storage then --checksum might

Hi there, I haven't yet delved into the rsync source code, and thus would need to ask a few questions first before going into the wrong direction. We have not, one but several cases[1] of doing Disaster Recovery type backup/synchronization with typically 1 to 2 million of files. Now yes, I know I can go and write a perl/python/<flavour_of_the_year> script to recursively do the

Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: -- SIP read from x.x.x.x:1024: REGISTER sip:mysipserver.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc From: "SPA922" <sip:5403@mysipserver.com>;tag=685bbad1fae3325do0 To:

Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows "Anwsering" but never does and the far end continues ringing until the voicemail answers, this then show as a disconnected call on the SPA922 I'm on the lastest firmware 6.1.5(a) Thanks in advance

Hi all, I've just bought some SPA922. First time with this hardware for me. I see no LAN tab in its web GUI where I can setup NAT for PC conected to its LAN ethernet port. However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way,

Hi there, I have a need to have scp pass the -P option to ssh to "bypass" the packetfilters that doesn't allow connections to return to arbitary "priviledged" ports, ie. ports <1024. See attached context sensitive diffs against 2.2.0p1 to please integrate. Thanx Hendrik Visage -------------- next part -------------- *** 1.1 2000/10/11 13:31:45 --- scp.c 2000/10/11

Conclusion: DO NOT USE SOFTWARE RAID-1 WITH EXT3 I've started with this conclusion cause i am positive that raid-1 and ext3 are causing data corruption. I've upgraded from 7.1 to 7.2 and from the first date i run into data corruption problems @ least once per day (i was monitoring once per day only) after 7 days we bought a brand new server cause the previous was very old and i thought

Hi all, I'm testing Linksys SPA922 phone and I have strange issue. when call is finished on the phone I see "CallEnded" and normal silence for cca. 5 seconds and then I get fast busy for cca. 20 sec. So, this isn't automatic hangup as on other phones I have tried (Cisco 7940, grandstream, XLite,... ) and I have to manually hangup handset to finish a call. Is this normal behavior

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches "fix" phones, the correspondant can't hear the caller. What

Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do

Hi, I am using ASTCC and trying to setup a calling card platform. The problem that I have is that astcc does not hangup calls correctly: 1. If I try to dial a number, call goes through fine. When I hang up the call from my side I get this: -- Called 192.168.1.56/1XX6872XXXX (masked a few digits) -- SIP/192.168.1.56-086c5000 is making progress passing it to

Hi All: I have experienced some big problems on an asterisk production server under 1.2.18: First of all, a very rare message like this... No application Macro ??? -- Saved useragent "Linksys/SPA922-5.1.7" for peer 1363 Jun 6 15:08:24 WARNING[406]: pbx.c:1720 pbx_extension_helper: No application 'Macro' for extension (pbx-incoming, 1133, 1) == Spawn extension

Is anyone aware of an milliwat/echo test number for telstra or similar? I want to fiddle a bit with gains but can't seem to get a hold of a miliwatt test number Im aware of a number they distribute for line quality testing for modems for BigPond, but I can't track it down .... -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au

Hello, I'm a newbie user of Asterisk, i'm sucessfully install it it's great but, i get some problems with echo in a adsl line. My system is a TDM440P with 3 FXO Ports and 1 FXS. Asterisk 1.2.13 Zaptel 1.2.11 Line 1 - Analog Line 2 - Analog with ADSL It's installed with two analog lines one of them have a ADSL splitter. The location are Spain, i load the tdm driver with opermode

Hi! I do a rsync between 2 machines. The throughput is only 2 MByte/Sec. Each machine is a Supermicro server with 2 x 8 Core Opteron 6128 64 GByte of ECC RAM 1 LSI MegaRAID SAS 9280-24i4e 24 x 2TByte SATA Disks as a RAID6 2 Intel Corporation 82599EB 10-Gigabit SFI/SFP+ Network-cards Both run Ubuntu 11.04 64Bit. Both use rsync version 3.0.7 protocol version 30 There are no

Database on a remote machine res_pgsql.conf: [general] dbhost=172.16.0.2 dbport=5432 dbname=ast_config dbuser=pbx dbpass= dbsock=5432 [Nov 28 20:12:02] DEBUG[6581] res_config_pgsql.c: Postgresql RealTime Host: 172.16.0.2 [Nov 28 20:12:02] DEBUG[6581] res_config_pgsql.c: Postgresql RealTime Port: 5432 [Nov 28 20:12:02] DEBUG[6581] res_config_pgsql.c: Postgresql RealTime User: pbx [Nov 28

On Wed, Apr 22, 2015 at 9:15 AM, Kevin Korb <kmk at sanitarium.net> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Normally, I would say that --checksum is actually slower than just > letting rsync re-copy everything Depends on the network capacity and costs associated with that bandwidth :( >and therefore is almost always the > wrong thing to do. Nope,