search for: tleilax

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123 password 123 default...

On 2/20/15 6:15 AM, thufir wrote: > What's the difference between user "123" and "devries"? Based on the > output here, they seem the same..? > > tleilax*CLI> > tleilax*CLI> sip show users > Username Secret Accountcode > Def.Context ACL Forcerport > 201 password 201 > default No Yes > 123 password 123 > default...

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at tleilax -m "hi" No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax Max-Forwa...

Hi, I **think** that I have user of thufir101, because I get a 200 response below, but I also get a 404. It seems to depend on how I send the ip address/fqdn? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes thufir101 password thufir101 default...

...ports for Asterisk to work even on a local level. so I'm just asking in general. For SIP to SIP peer calling, and by that I just mean "ring" or "beep," some sort of ping, basically, just configure the two softphones to use the IP address for the Asterisk box? also: tleilax:~ # tleilax:~ # asterisk -V Asterisk 1.8.32.1-vici tleilax:~ # tleilax:~ # asterisk -rm Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for...

On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: > This is showing nothing so I don't think your test message even made it > here. I think it looped in the 'doge' server. I was wondering the same thing :) in tleilax, I looked in /var/log/asterisk/messages and see: [Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19] <--- SIP read from UDP:192.168.1.3:38154 ---> OPTIONS sip:345 at tleilax SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:38154;branch=z9hG4bK.77fd156e;rport;alias From: sip:sipsak at 127.0.1.1:...

On 2/20/15 2:29 PM, thufir wrote: > On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > > >> A "sip set debug on" will give you more info on why you are getting the >> 404. It probably has to do something with your context/dialplan. > > on tleilax: > > tleilax*CLI> > tleilax*CLI> sip set debug on > SIP Debugging enabled > tleilax*CLI> This is showing nothing so I don't think your test message even made it here. I think it looped in the 'doge' server. > > > on doge: > > thufir at doge:~$ &g...

...gt; > > > so I'm just asking in general. For SIP to SIP peer calling, and by that > I just mean "ring" or "beep," some sort of ping, basically, just > configure the two softphones to use the IP address for the Asterisk box? > > > also: > > > tleilax:~ # > tleilax:~ # asterisk -V > Asterisk 1.8.32.1-vici > tleilax:~ # > tleilax:~ # asterisk -rm > Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; ty...

I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX...

.... While Skype works (mostly) from my LAN, the connection isn't the greatest. My LAN uses a wireless bridge to connect to another LAN. It's just a home setup; it is what it is. How do I test a connection? How do check the settings? As far as I can tell, the settings are correct. tleilax:~ # tleilax:~ # asterisk -V Asterisk 1.8.32.1-vici tleilax:~ # tleilax:~ # asterisk -rm Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' fo...

...dyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Sun, Feb 22, 2015 at 8:25 AM, thufir <hawat.thufir at gmail.com> wrote: > I'm looking into the dialplan specifics: > > tleilax:~ # > tleilax:~ # cat /etc/asterisk/extensions.conf > [general] > static=yes > writeprotect=no > > [globals] > CONSOLE=Console/dsp ; Console interface for > demo > TRUNK=DAHDI/r1 ; Trunk interface > TRUNKX=...

..._server_ip.pl Official paid-for ViciDial support is available at http://www.vicidial.com Free community-based ViciDial Support is available at http://www.vicidial.org/VICIDIALforum - ViciBox Redux v.6.0.3-141118 Could not chdir to home directory /home/thufir: No such file or directory thufir at tleilax:/> thufir at tleilax:/> nmap 192.168.1.3 Starting Nmap 6.40 ( http://nmap.org ) at 2015-02-18 09:14 EST Nmap scan report for 192.168.1.3 Host is up (0.00075s latency). Not shown: 998 closed ports PORT STATE SERVICE 22/tcp open ssh 2000/tcp open cisco-sccp Nmap done: 1 IP address (1...

...20/15 3:20 PM, thufir wrote: > On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: > >> This is showing nothing so I don't think your test message even made it >> here. I think it looped in the 'doge' server. > > I was wondering the same thing :) > > > in tleilax, I looked in /var/log/asterisk/messages and see: > > [Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19] > <--- SIP read from UDP:192.168.1.3:38154 ---> > OPTIONS sip:345 at tleilax SIP/2.0 > Via: SIP/2.0/UDP 127.0.1.1:38154;branch=z9hG4bK.77fd156e;rport;alias > Fr...

...am having a look at llvm-gfortran; it compiles OK, but I do not know how to using the JIT as described in http://llvm.org/docs/GettingStarted.html#tutorial. With llvm-gcc I managed to invoke the lli properly, but with llvm-gfortran compiled programs, it fails with the following message: philou at tleilax:~/gcc/llvm/install/bin$ lli ./main.bc 'main' function not found in module. I understand it does not find the entry point of the program ( gfortran creates a MAIN__ function to be linked with some other code ). What should I so to have the lli running fortran code ? Regards, Philippe...

...(i386/linux)) Content-Length: 0 ** reply received after 161.445 ms ** SIP/2.0 483 Too Many Hops final received thufir at doge:~$ but that's ok. How can I test, I mean make a voice call, given that I only have two computers to work with at the moment? The server runs Asterisk on tleilax, and doge is the client. Both connect to the same router. the ip address for tleilax is 192.168.1.2 and the ip address for doge is 192.168.1.3 (generally; doge uses DHCP). When I get more pc's, I can maybe have doge call another pc on the network, but, for right now, what can I do to test...

I called 102 from 101 successfully! I have everything connected to my home router. Asterisk is running on tleilax, so I used my Android phone to call doge. Worked like a charm. I'd been thinking that the firewall was blocking connections, but not at all. Anyhow, thanks to everyone who's help me out. I'm sure I'll have other problems, but huge milestone. -Thufir