similar to: Console latency

Hello, I am trying to connect a instance of ices2 to icecast2. It seems that the connection is done well but I do not know how to take the microphone input to send it to the server. Any help is highly appreciated. Manuel The information is this: Part of my ices.xml: <input> <module>alsa</module> <param name="rate">44100</param> <param

hi list. after my unsuccessfull experiences with mISDN i tried again to implement a zaphfc based solution. problem is: sound on calls via capi is stuttering/broken and therefore unuseable. my conf: - cel 1300, 256mb ram - avm b1 via capi connected to my outgoing ISDN - acer surf pci via zaphafc and crosslink cable with termination as internal bus - siemens gigaset 3035 as internal phone -

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

Hello, I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring but I don't hear the caller and the caller doesn't hear me (all IP Phones have the same problem). This problem appear also if the call is directly send to the second E1 of the digium card who is connected to an IVR. It does not depand on the charge of the server (I have the problem with only one call).

My SIP seems to be configured correctly as I can dial out and my minutes show up on my broadvoice bill, but whenever anyone calls my broadvoice # inbound they just get a busy signal. I dont get anything in the logs saying anything came in from broadvoice at all. Has anyone had this/simmilar problem with inbound from Broadvoice? Any suggestions? Thanks Neri -------------- next part

Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal.

Hello We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since. We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: It does show up in asterisk a few seconds after the UIP200 reboot: -- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200 but

Hi, People here often work on 2-3 places (office 1, office 2 and home). I would like to give them 1 extension (XXX) and to ask them to 'register' the phone they use at a certain moment. The idea is that, when you need someone, just dial XXX and the phone near him (in Office 1, Office 2 or at Home), will ring. This will keep my queue system and other tricks intact, where I always use the

Hello I just set up Asterisk 1.6.2.9 through packages on a test host running Ubuntu 11.04, configured sip.conf/extensions.conf, and launched EyeBeam 1.5.20 to run the echo test. For some reason, even through I'm using a headset, there's a lot of echo and after a few seconds, it sounds like it enters a very fast loop before the echo stops somewhat. IOW, unusable sound. Here's a

> -------------------------from aoTuV_b3_En.txt----------------- > Manuke's patch is used for improvement in the speed of sort > processing > (thank you!). > -------------------------------------------------------------- http://www.geocities.com/Area51/Dungeon/6809/vorbis-optsort-en.html This patch accelerate encoding time about 10%. Why not include in 1.1.2?

> -------------------------from aoTuV_b3_En.txt----------------- > Manuke's patch is used for improvement in the speed of sort > processing > (thank you!). > -------------------------------------------------------------- http://www.geocities.com/Area51/Dungeon/6809/vorbis-optsort-en.html This patch accelerate encoding time about 10%. Why not include in 1.1.2?

ok.. game is - "Ground Control 2" error in console during install is as follows.. > err:ole:TLB_ReadTypeLib Loading of typelib L"C:\\Program Files\\Common Files\\InstallShield\\Professional\\RunTime\\IsProBE.tlb" failed with error 2 > it freeze's...

Please excuse my ignorance as I''m a linux newbie. Basically I have a setup of an adsl ethernet modem (nated and then everything forwarded to the external ip of my Mandrake mnf firewall) connected to the mnf firewall which then connects to the lan. internet <--> adsl modem <--> mnf firewall <--> lan There''s only 2 nics in the mnf firewall so it''s a

Hi, I currently have a E100P card installed on my machine and the E1 subscription will be activated pretty soon. However, I have no idea how to configure asterisk to make inbound and outbound call using the E1. Especially for extensions.conf. Below is the configuration I used for zaptel.conf and zapata.conf. Is it possible if someone can verify if the configuration for zaptel and zapata is

I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context: [callback] exten=> 501,1,Congestion() exten=> 501,2,Hangup() exten =>h,1,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =>h,2,Hangup() With the above, the call comes into the trigger number, then the call

We have tried Asterisk 1.0.9 on FC4 and have never been able to get CAPI (with Fritz card, fcpci) to work properly. Apart from that Asterisk works fine in switching internal calls. But it's useless if we can't make outgoing calls on our ISDN line. We are considering abandoning FC4 for Debian or SuSe. What is the general concensus on the best Linux to run Asterisk with CAPI? /Why Tea

Hello, I''m running mongrel_cluster and have been using it successfully for quite a while now. However, today I ran a "cap deploy" and started seeing the following error: !!! PID file /var/www/apps/jco/shared/log/mongrel.8013.pid does not exist. Not running? for each of the mongrels that should start up. I''ve googled for solutions but haven''t found any

On Sunday 21 August 2016 15:32:55 Frank Cox wrote: > On Sun, 21 Aug 2016 14:55:50 -0700 > > j_post wrote: > > On Sunday 21 August 2016 13:50:25 Frank Cox wrote: > > > On Sun, 21 Aug 2016 13:39:29 -0700 > > > > > > j_post wrote: > > > > Sound stopped working on my CentOS 6.8 system several months ago. > > > > > > alsamixer may

Hi, I just transformed an old Pentium III 500 into a headless jukebox. It's installed in the basement, near the stereo. There's only a base CentOS system on it (GNOME unchecked, package customization checked and then everything unchecked). From there on, I just installed the ALSA utils, and vorbis-tools. The machine is only supposed to do one thing (and to do it well, UNIX philosophy

>Hi, > >I just transformed an old Pentium III 500 into a headless jukebox. It's >installed in the basement, near the stereo. There's only a base CentOS >system on it (GNOME unchecked, package customization checked and then >everything unchecked). From there on, I just installed the ALSA utils, >and vorbis-tools. The machine is only supposed to do one thing (and to