search for: mediaport

...ng from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in context 'smvoice-mediaport'. When doing the "dialplan show" it clearly in the context. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' => 1. Goto(smvo...

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1104' => 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'mediaport_direct' => 1. Goto(smvoice-mediaport-pu...

I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. Is there a particular parameter needed for 1.6 that 1.4 did not care about? If I drop back to 1.4 it starts working again. Thanks Jerry

...> > On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote: > > > > >/ <------------> > > />/ ?[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: > > / Call from 'devcentos5x64_to_ebox4300' to extension > > 'mediaport_audio_visual' rejected because extension not found. > > > > Jerry-- > > > > from the console, type "dialplan show smvoice-mediaport", and > > let's verify for certain that it's in there. > > > > I'll try to reproduce your problem i...

...es the same thing exist in ALSA??? some traces below Jerry == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.26.1 currently running on ebox3850 (pid = 3902) Verbosity was 0 and is now 5 -- Executing [mediaport_audio_visual at smvoice-mediaport:1] Goto("SIP/devcentos5x64_to_ebox3850-0837c170", "smvoice-mediaport-audio-visual|s|1") in new stack -- Goto (smvoice-mediaport-audio-visual,s,1) -- Executing [s at smvoice-mediaport-audio-visual:1] ChanIsAvail("SIP/devcentos5x64...

1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all

Daniel, I don''t know if you found a solution to your problem, I''m currently passing a DVB-S card through to my XPSP3 DomU with Mediaportal as the streaming app. Anyhow, I was having performance issues with my mpeg2 streams as well. I ended up doing the following on my Q6600: xm vcpu-set Domain-0 3 xm sched-credit -d DomU -w 2048 With the Dom0 pinned to 3 cpus, and my DomU on the fourth, and the creditscheduler set much higher on...

...7,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0 RingBackTone: 2,30831,30467,1943,2111,0,16000,32000,0,0,0 CallWaitTone: 1,30831,0,5493,0,0,2400,2400,4800,0,0 AlertTone: 1,30467,0,5970,0,0,480,480,1920,0,0 NPrintf: 0.0.0.0.0 TraceFlags: 0x00000000 SyslogIP: 0.0.0.0.514 SyslogCtrl: 0x00000000 MediaPort: 16384 CFGID: 0x00000000 ata0013199e70f5 Version: v3.1.1 atamgcp (Build 040629A) Features: 0x00000017 HardwareVersion: 0x0010 0x0000

Super squirrel Here is a nice ad with "the squirrel" http://www.mediaportal.ro/play.php?pid=13539&Super%20Squirrel%20Scream --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFFw@p...

My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k on Debian stable). It could, however, register with another installation of Asterisk and the settings on the phone (apart from the SIP proxy address) haven't changed since then. On the new Asterisk box my sip.conf contains this: [jeremy] type=friend regexten=801 allow=g729 host=dynamic secret=PASSWORD nat=yes

...ayer ./mn_install.sh # Install cepstral before MessageNet cd /mnt/silentmhome/cepstral ./mn_install.sh # Install the latest version cd /mnt/silentmhome/Versions ./mn_install.sh # Set path location source /etc/profile ## Setup RAID arrays for booting #/home/silentm/bin/smset_raidboot.sh # Setup Mediaport #/home/silentm/bin/smclient.sh -multimediapc PROC_CMD_LINE_MN_KICKSTART=`grep -c "mn_demobox" /proc/cmdline` if [ "$PROC_CMD_LINE_MN_KICKSTART" = "1" ] then /home/silentm/bin/smfunctions -set_node_name demobox echo demobox | passwd --stdin else # Default this mach...

Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??