similar to: Troubles with Kphone

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

Dear Asterisk experts, I've just downloaded Asterisk CVS version (since I'd like to manage its configuration from RealTime). Next, I have kphone on the same Linux host, and VmWare virtual machine with Windows and X-Lite IP phone inside. I successfully tested the demo's with X-Lite, but failed to hear something with kphone (v4.1.1). There were NO problem with this kphone and stable

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

Hello, I know that my Kphone question may be a bit off topic, but I have been busy with this again and again for about one month now, sent three mails to kphone@wirlab.net (the contact address mentioned on http://www.wirlab.net/kphone/index.html), asked for a solution in a german ip phone forum and tryed many things by myself. I try to compile KPhone 4.0.3 (tryed CVS Version as well) but

I need to redirect the sound from a remote Centos 5.2 computer to my local Centos 5.2 computer. Both are i386 OS. Searching the web and Centos web site has indicated that it is possible but I have not found any information about how to do it. I am currently using ssh and/or vnc to display the remote computer locally. At this point, the sound is being played on the remote computer only.

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

I am experiencing a strange problem and wanted to know if someone has faced any similar issues or could provide me with a way to counter this problem. I am in the process of experimenting with asterisk and trying to setup a basic functional system. I have one TDM400P (single port) and one X100P. I am using one analog phone connected to the TDM400P and I also have a couple of Xlite SIP phones

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

When I try to make a call, I have this error: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it):

* Enrico Minack (enrico.minack@informatik.tu-chemnitz.de) écrivait : > > I use kmix as mixer, but there is nothing in it about a > > "capture channel". How could I find where it is defined ? > then try alsamixer or amixer and watch out for capture and unmute and apply > this for the according channel (mic, line-in, pcm or master) Hum... the problem is that RH 9 uses

Hello Mark, >> What is known about Nvidia plans - do they plan to develop a dom0-only, >> or maybe even a domU drivers as well? > dom0 3D drivers are certainly easier (these should also work in a domU that > has access to PCI graphics card, once PCI-passthrough is working again - > someone will have to figure out how to make X behave correctly, tho). Ooh, that sounds

When I''m executing cucumber tests, I noticed that sometimes rails app (in test env.) getting several the same requests (GET or POST) usually around 3, and it doesn''t render anything with empty HTTP status code. Have anyone met something similar to that issue? here is some example of log file: Started POST "/account" for 127.0.0.1 at 2010-11-30 22:34:17 +0200

How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? /HHA

I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. ------------------------------- [chan_alsa.so] => (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20

Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could possibly be wrong? If it were interrupt overload, I'd hear lots of cracks in my echo, right? I'm not hearing that. Besides, a telephony card is not involved.

* EvilOverlord (eviloverlord@kucs.net) écrivait : > In whatever you use as mixer control for the soundcard (alsamixer, > aumix, etc) what is set as the "capture" channel? Your soundcard may > not support capturing what is being played. If it helps : my soundcard is a ES1988 Allegro-1, the module is "maestro3". I use kmix as mixer, but there is nothing in it about a

Potentially nasty bug in current HEAD when printing the name of the file being played (the "Playing: filename.ogg" bit), any % symbols in the name appear to be printf parsed causing either junk to be printed or a crash or both. I noticed this when trying out http streaming as the url has %20's in it for spaces in the file name. Oh, and it still says it's rc2... --

I would like to setup my Asterisk server to process an incoming SIP URI and redirect all requests to a specific context. Example: (1) using a sip phone I'd like to be able to call: sip:somedomain.com *or* sip:someone@somedomain.com (2) i'd like my asterisk server to answer the call and route it to the context=in-from-sipclient which would play thru some DP actions Can anyone give

Hello Folks! in my sip-logs i see that asterisk uses the User-Agent ID "Asterisk PBX": SipClient: Received: 16:34:03.023 --------------------------------- BYE sip:102141@131.130.XXX.XXX:44343;transport=udp SIP/2.0 Max-Forwards: 10 Record-Route: <sip:213.2XX.XXX.XX8;ftag=as2eb3c466;lr=on> Via: SIP/2.0/UDP 213.2XX.XXX.XX8;branch=z9hG4bK539a.47e6e8a7.0 #this is SER Via:

Hi list, I bought two UTStarcom F1000 phones, pre-equipped with the latest firmware, including WPA support. Those are configured to register to an asterisk server on the internet (not LAN), and registration works. Calling and being called also, with transfer and all bells and whistles. After a few minutes up to 5 hours (varies widely), the display tells me that an Accesspoint is not available