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On 25 May 2004 at 9:50, sdgesa gaeharth wrote: > I was using windows media player (with the patch for playing ogg). You mean you're using it for ogg-streaming? If I remember correctly just a few mails ago somebody said it only plays static files. Which plugin (patch???) are you using? Regards, Stefan --- >8 ----...

There seems to be a long delay from wen I speak into the mic on a broadcast server and when I hear the speaking on the users end. Can anyone explain this? I am using the latest versions of ices and icecast. Ices is on a different server that icecast and at another location. Thanks --------------------------------- Do you Yahoo!? Friends. Fun. Try the all-new Yahoo!

Seems here is the right place? http://mediaxw.sourceforge.net/ What is a bit strange: It claims to have had no updates since 2001 (MediaXW-package itself in the SF-downloadsection) / 2002 (website)? <p> Stefan On 25 May 2004 at 17:51, sdgesa gaeharth wrote: > MediaXW. I dont remember where i got it... > > > --- Stefan Neufeind <stefan@neufeind.net> wrote: > > On 25 May 2004 at 9:50, sdgesa gaeharth wrote: > > > > > I was using windows media player (with the patch > > for playing ogg). &g...

Message: 12 Date: Tue, 17 Oct 2006 18:07:04 -0700 (PDT) From: sdgesa gaeharth <pollux1234567890@yahoo.com> Subject: Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute To: asterisk-users@lists.digium.com Message-ID: <20061018010704.55118.qmail@web50806.mail.yahoo.com> Content-Type: text/plain; ch...

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

> I dont care I just wanted to make sure there is no problem 50 seconds is a bit long. Probably your client buffer is really high and you are streaming at a really low bitrate. But even in that case 50 seconds seems long. If you can post more details on bitrate, client used, buffer size used on the client, etc, that would be helpful for verification. jack. --- >8 ---- List archives:

..., and using the sample configuration files with that firmware to set up your phone. This SHOULD work. If it still does not work after doing this, there may be a hardware issue with your phone. - Noah ------------------------------ Message: 14 Date: Thu, 16 Mar 2006 10:53:22 -0800 (PST) From: sdgesa gaeharth <pollux1234567890@yahoo.com> Subject: Re: [Asterisk-Users] Re: transfers/parked calls + polycom 501 To: asterisk-users@lists.digium.com Message-ID: <20060316185322.94183.qmail@web52609.mail.yahoo.com> Content-Type: text/plain; charset="iso-8859-1" I am using the late...

On Tue, May 25, 2004 at 08:53:56AM -0700, sdgesa gaeharth wrote: > > There seems to be a long delay from wen I speak into the mic on a > broadcast server and when I hear the speaking on the users end. Can > anyone explain this? I am using the latest versions of ices and > icecast. Ices is on a different server that icecast and...

Hi, We run a small switchboard using Asterisk and Free PBX. We have two main extensions and two ring groups. The first ring group rings the two internal extensions. If the internal extensions do not pick up the call after 15 seconds then the second ring group kicks in which should ring the two internal extensions plus two external numbers. Firstly, how do I pass the DID number of an incoming

I need to set up an icecast server were about 100 people are going to be listining. The ices stream is going to be sent from one location to the icecast server at another location. The icecast server is 3 GHz and 512 of ram. Bandwidth is 100 MB/s. thanks <p> __________________________________ Do you Yahoo!? Yahoo! Domains – Claim yours for only $14.70/year

will the new version of ices/icecast support sending live broudcast sent as mp3 as opposed to ogg. I remember su ccesfullty sending an mp3 song over the net but I dont remember what happened with a live broadcast. thanks <p> __________________________________ Do you Yahoo!? Yahoo! Domains – Claim yours for only $14.70/year

I was using windows media player (with the patch for playing ogg). <icecast> <limits> <clients>1000</clients> <sources>10</sources> <threadpool>5</threadpool> <queue-size>102400</queue-size> <client-timeout>30</client-timeout>