[Asterisk-Users] Zyxel P2000W web interface?

2004-10-12 Thread Andreas Mikkelborg
Hey,
I've got this phone, and it's working allmost perfectly ;p

Firstly I get some unknown codec errors in asterisk so the first half 
second when I make a call is blurry, after that it's just fine.
Then it's that menu selections won't work.
According to www.voip-info.org/wiki-Asterisk+phone+Zyxel+P2000W

I can configure DSP settings in the web interface, but the web interface 
I get there's no such option, nor on the actual phone, anyone know of 
any clever workarounds?
Thanks.

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Re: [Asterisk-Users] Status of conference calls at Astricon ?

2004-09-22 Thread Andreas Mikkelborg
On 14:48 Wed 22 Sep , Matteo Brancaleoni wrote:
> /me too
> this morning was all okie, now I can't connect.
> 
> I have an asterisk server ready for replicate the conference
> here in .it, as soon as the link will be up with someone,
> I'll post the IAX2 url

How can I set it up to replicate,so I can help to, hm, although this 
is just a 1Mbit ADSL though :P


> -- 
> 
> Matteo Brancaleoni
> System Administrator
> [EMAIL PROTECTED]
> 
> EspiA Srl - e*solution provider
> Via Pascoli, 37
> 20129 Milano - Italy
> SIP:[EMAIL PROTECTED]
> Tel. +39 0270633354
> Fax. +39 0245487890
> IAXTEL: 17005662458
> http://www.espia.it
> 
> 
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Re: [Asterisk-Users] Status of conference calls at Astricon ?

2004-09-22 Thread Andreas Mikkelborg
On 05:36 Wed 22 Sep , Jeremy McNamara wrote:
> Matteo Brancaleoni wrote:
> 
> >so, bring on and demostrate to the world
> >what asterisk can do!
> 
> I have packed all the necessary gear to stream the Developers Meeting on 
> Friday.  I am looking for people with Big Pipe(tm) to get crazy and link 
> multiple Asterisk MeetMe's together.  Lets see how much Pipe she can take.
> 
> ; normal usage until it is abused with MOH!
> IAX2/[EMAIL PROTECTED]/4569
> 
> ; if you plan to link in your own MeetMe use this
> IAX2/[EMAIL PROTECTED]/4569|qm  ; Monitor only and be quiet!

So I can just call that "nr" and listen to the meeting on Friday? (what 
time?)
I just added that line to extensions.conf, but now I just get no answer 
:P

-- Executing Dial("SIP/3000-157d", "IAX2/[EMAIL PROTECTED]/4569") in new stack
-- Called [EMAIL PROTECTED]/4569
-- Call accepted by 66.225.202.72 (format GSM)
-- Format for call is GSM
-- Hungup 'IAX2[astricon-in]/1'
  == No one is available to answer at this time


> 
> Then to truly experience Asterisk's power, join the same conference...
> 
> ...Using SIP: [EMAIL PROTECTED]
> 
> ...and using H.323: [EMAIL PROTECTED]
> 
> However, you can also call 877-677-9648 using room number 4569 to get 
> into the same conference, if you are not 'leet enough to have any VoIP 
> skills.
> 
> Feel free to abuse the system as necessary. My thugs are ready to break 
> knee caps if someone starts playing games.
> 
> There have been a few different ideas thrown around on how to integrate 
> the participants in the MeetMe to the developers meeting,  I will 
> discuss this with others when the time is appropriate. Hopefully we will 
> be able to take questions from the MeetMe, so start pondering some good 
> questions.
> 
> If the powers that be let me stream more content to the MeetMe, I will. 
> So you better hang out in the conf all day, what else do you have to do, 
> work?
> 
> 
> A few random numbers to call:
> 
> Twisted has lit up 866-251-9527 to do various crazy things, feel free to 
> call him, any time.  :)
> 
> Then 866-498-4988 and 248-724-1710 are routed to my mobile Asterisk box, 
> if it has 'net access. Be prepared to deal with IVRs to find a human.
> 
> Then, if you are really bored you can call 877-966-6673, but don't 
> expect to talk to a human and you will most likely hangup in disgust, so 
> maybe you shouldn't call this one.
> 
> Give us a few hours and we will think of some more crazy things to do.
> 
> 
> 
> Jeremy McNamara
> 
> 
> 
> 
> 
> 
> 
> 
> p.s. So, I left out MGCP and SCCP, sue me, my legal department is bored
> 
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