similar to: Call failed: 408 timeout

Displaying 11 results from an estimated 11 matches similar to: "Call failed: 408 timeout"

2011 Oct 21
3
high-privileges
hello all, my vm (windows server 2008) keep crashing? 2011-09-21 15:00:45.924: 1355: info : libvirt version: 0.9.6 2011-09-21 15:00:45.924: 1355: error : virSysinfoRead:465 : internal error Failed to find path for dmidecode binary 2011-09-21 15:05:43.873: 1349: warning : qemuDomainObjTaint:1128 : Domain id=1 name='IIS' uuid=a02e41a9-9f4d-d94f-43a6-995b1ca6d37d is tainted:
2009 Mar 20
0
Asterisk Realtime Configuration and 404 Extension not found
Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |
2009 Mar 24
0
Asterisk Realtime Config and SIP/401 Unauthorize: why?
Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context | secret |
2009 Mar 19
0
Extensions not found and 401 Unauthorized in realtime configuration (Long post)
Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |
2005 Oct 13
1
Noob help with IAX
Ok so I've just built and installed a CVS (HEAD) version of asterisk on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples via "make samples". Everything seems to work except one thing. I'm trying to do the connect to the Digium IAX demo server portion of the demo (dial 500) and I just get the following messages. I am behind a NAT server and did NOT change
2003 Mar 21
1
PXELinux can
Hi Josef, I have included the hex dump, I hope it will help to pinpoint the problem. "Josef Siemes" <jsiemes at web.de> wrote: >Hi, > >> I got tftp-hpa to compile on the IRIX box, but it doesn't >> work any better. Any ideeas on what is wrong, and >> how can I fix it? > >The dumps are to few for that. Give a hex dump of the >TFTP request,
2007 Feb 14
0
Asterisk & CME integration using h323
Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323. Cisco conf: dial-peer voice 8 voip destination-pattern 2... session target ipv4:<asterisk ip> codec g711alaw no vad h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ;tos=lowdelay ; disallow=all allow=alaw allow=ulaw allow=gsm context=from-internal extension.conf [from-internal] exten =>
2005 May 26
1
Asterisk con X-lite : Register Ok but no calls (404 Not found)
Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer.
2005 Jul 27
0
Please, I looking for halp!
Hi all! I don't cnow where can i ask about this and I hope you can halp me. I have a digital voice recorder olymus ds-2300, it's working with *.dss files (digital speech standart). There is Olympus Dss Player for Windows but nathing can play it with linux. On olympus development homepage i found "Sound SDK for WIndows" but they don't wont to make it open. Please can you
2009 Dec 07
1
Error : SIP/2.0 401 Unauthorized
Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathelper.so"
2006 Mar 06
1
Wins Installation problem on solairs x86 version 5.10
Follwoing errors are shown while installation. Any quick help will be highly appriciated. config.c: In function `init_server_dir': config.c:217: warning: right shift count >= width of type config.c: In function `init_paths': config.c:256: warning: unsigned int format, uid_t arg (arg 3) mmap.c: In function `try_mmap_fixed': mmap.c:107: warning: implicit declaration of function