Displaying 9 results from an estimated 9 matches for "pollux1234567890".
2006 Oct 18
1
Windows and file shares
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; charset="iso-8859-1"...
2006 Mar 16
1
Re: transfers/parked calls + polycom 501
...le
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 latest firmware and bootrom and th...
2006 Mar 03
4
really need help with outgoing calls..PSTN errors
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
2006 Jun 22
4
when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on
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
2006 Jan 31
3
ZAP <--> sip(polycom301) can not hear each other
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
2004 Aug 06
2
100 users; icecast capable?
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>
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2004 Aug 06
2
mp3 for live stream
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>
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2004 Aug 06
0
long delay
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>
2004 Aug 06
5
long delay
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
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