[Asterisk-Users] french newbie with asterisk
kim
kimbo at nerim.fr
Thu Oct 16 00:31:11 MST 2003
asterisk-users-request at lists.digium.com wrote:
>Send Asterisk-Users mailing list submissions to
> asterisk-users at lists.digium.com
>
>To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
>or, via email, send a message with subject or body 'help' to
> asterisk-users-request at lists.digium.com
>
>You can reach the person managing the list at
> asterisk-users-admin at lists.digium.com
>
>When replying, please edit your Subject line so it is more specific
>than "Re: Contents of Asterisk-Users digest..."
>
>
>Today's Topics:
>
> 1. Re: Digium should develop and sell just Dummy card. For timing... (Adam Hart)
> 2. Re: My Grandstream works,
> but my X-Lite doesn't:no sound after 5sec (rnc Info Lists)
> 3. Re: Wildcard TDM400P - FXO? (Steve Meyers)
> 4. e100p in Australia (Stephen Dredge)
> 5. Re: DISA and ringing tone (John Todd)
> 6. Re: My Grandstream works, but my X-Lite doesn't:
> no sound after 5sec (WipeOut)
> 7. Re: Digium should develop and sell just Dummy card. For timing... (Tilghman Lesher)
> 8. Asterisk capacity (vincent nguyen)
> 9. Re: My Grandstream works, but my X-Lite doesn't:
> no sound after 5sec (WipeOut)
> 10. Re: My Grandstream works, but my X-Lite doesn't:no sound after 5sec (Paul Cheng)
> 11. Re: e100p in Australia (Anthony Wood)
> 12. Re: E100P setup in Switzerland (Marcel Prisi)
> 13. Re: 200-400ms latency (Olaf Menzel)
> 14. frensh newbie with asterisk (kim)
>
>--__--__--
>
>Message: 1
>From: "Adam Hart" <adam at teragen.com.au>
>To: <asterisk-users at lists.digium.com>
>Subject: Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
>Date: Wed, 15 Oct 2003 14:57:55 +1000
>Reply-To: asterisk-users at lists.digium.com
>
>From: "Anton Tinchev" <atl at unixsol.org>
>To: <asterisk-users at lists.digium.com>
>Sent: Wednesday, October 15, 2003 1:06 PM
>Subject: [Asterisk-Users] Digium should develop and sell just Dummy card.
>For timing...
>
>
>
>
>>I'm first to buy 5 pack. Even for > $30.
>>
>>
>>
>
>Doesn't ztdummy already do this?
>
>
>--__--__--
>
>Message: 2
>Date: Wed, 15 Oct 2003 07:00:43 +0200 (CEST)
>Subject: Re: [Asterisk-Users] My Grandstream works,
> but my X-Lite doesn't:no sound after 5sec
>From: "rnc Info Lists" <info-lists at robertc.de>
>To: asterisk-users at lists.digium.com
>Reply-To: asterisk-users at lists.digium.com
>
>Do you have a 100 or 101? You have indicated different models in your
>postings. Were you able to get Call Transfer and Call Waiting working
>with your Asterisk system and other phones? Which version of the
>Grandstream firmware do you use? There most recent on their website this
>weekend was at least 2 version numbers higher than what came on my phone
>in August. Think that they are making improvements rather frequently.
>
>
>Robert
>
>
>
>
>>On Wed, 15 Oct 2003, Jon Pounder wrote:
>>
>>
>>
>>>>The Grandstream 101 I'm using is a piece of junk but I don't have the
>>>>
>>>>
>>>same
>>>
>>>
>>>>problem with it.
>>>>
>>>>
>>>What don't you like about the grandstream ? (I am not looking to flame
>>>you,
>>>but was considering buying and if there are problems would rather find
>>>out
>>>beforehand)
>>>
>>>
>>Nothing works. Call transfer and call waiting, in particular. (Well,
>>almost nothing; vm notification does work)
>>
>>There is no place to plug in a headset, and since I do a fair amount of
>>tech support and longish conference calls, that's a big deal for me.
>>
>>However, keep in mind that I have an old, no-longer-manufacturered model
>>(the Budgetone 100). Don't take my frustration with my outdated phone as
>>a sign that you should dismiss Grandstream out of hand - I just don't like
>>my 100.
>>
>>--
>>JustThe.net Internet & Multimedia Services
>>22674 Motnocab Road * Apple Valley, CA 92307-1950
>>Steve Sobol, Proprietor
>>888.480.4NET (4638) * 248.724.4NET * sjsobol at JustThe.net
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>
>
>--__--__--
>
>Message: 3
>Subject: Re: [Asterisk-Users] Wildcard TDM400P - FXO?
>To: Asterisk List <asterisk-users at lists.digium.com>
>Date: Tue, 14 Oct 2003 23:27:21 -0600
>From: Steve Meyers <steve-asterisk at spamwiz.com>
>Reply-To: asterisk-users at lists.digium.com
>
>On Tue, 2003-10-14 at 19:31, Gene Kochanowsky wrote:
>
>
>>Does anyone know if or when the FXO daughter boards for the TDM400P will be available?
>>
>>
>
>September 2003. :)
>
>--__--__--
>
>Message: 4
>Date: Wed, 15 Oct 2003 12:50:33 -0500
>From: Stephen Dredge <asterisk at wowinternet.com.au>
>To: asterisk-users at lists.digium.com
>Subject: [Asterisk-Users] e100p in Australia
>Reply-To: asterisk-users at lists.digium.com
>
>
>I've seen this question asked before but haven't seen a definative answer.
>Does the e100p work in australia? Did any one who was asking the question
>before bite the bullet and get one? I can get a te410 if i really have to but
>would prefer to stay with the cheaper option.
>
>Any comments appreciated, Thanks
>
>
>-------------------------------------------------
>This mail sent through IMP: http://horde.org/imp/
>
>
>--__--__--
>
>Message: 5
>Date: Tue, 14 Oct 2003 23:05:19 -0700
>To: asterisk-users at lists.digium.com
>From: John Todd <jtodd at loligo.com>
>Subject: Re: [Asterisk-Users] DISA and ringing tone
>Reply-To: asterisk-users at lists.digium.com
>
>
>
>>Hi
>>I am using DISA to get my Polycom SoundPoint400 with H323 firmware
>>to connect to *
>>I have it working, but when I dial SIP end points there is no
>>ringing tone on the phone. DISA gives dial tone but does not give
>>ringing (if I understand correctly it is because it expects to
>>transmit sound created by terminating side of the call)
>>Is there a way to make DISA application to generate ringing tone
>>back to the handset of the originating end point?
>>
>>Thanks,
>>Serge
>>
>>
>
>Serge -
> I don't know about H323, but I get ringtones from DISA on a SIP
>outbound channel. Try adding "r" to the options list on your Dial
>statement.
>
>JT
>
>--__--__--
>
>Message: 6
>Date: Wed, 15 Oct 2003 07:46:52 +0100
>From: WipeOut <wipe_out at lycos.co.uk>
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:
> no sound after 5sec
>Reply-To: asterisk-users at lists.digium.com
>
>Steven J. Sobol wrote:
>
>
>
>>X-Lite build 1079 consistently chokes no matter which codec I use -
>>after five seconds I suddenly have no sound coming in and possibly no
>>sound going out too. Putting the line I'm on on hold and then switching
>>back to it gives me another five seconds of sound, then it dies, etc.
>>
>>
>>
>If you are not using a headset then X-Lite cuts the sound after a few
>seconds.. Its probably the echo cancelation kicking in.. Connect a
>headset and all should be fine..
>
>Later.
>
>
>--__--__--
>
>Message: 7
>From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] Digium should develop and sell just Dummy card. For timing...
>Date: Wed, 15 Oct 2003 01:48:56 -0500
>Reply-To: asterisk-users at lists.digium.com
>
>On Tuesday 14 October 2003 23:50, Chris Albertson wrote:
>
>
>>If the software needs a specialcard to keep time then the
>>software is broken or poorly designed.
>>
>>
>
>Don't complain so loudly unless you're willing to contribute the
>fixes. Opinions are like assholes, and you know where that's going.
>Takes something else entirely to fix a perceived problem.
>
>If the system is so horribly broken, why are you using it as-is and not
>fixing it?
>
>-Tilghman
>
>
>--__--__--
>
>Message: 8
>From: "vincent nguyen" <vhvincent at hotmail.com>
>To: asterisk-users at lists.digium.com
>Date: Wed, 15 Oct 2003 16:51:42 +1000
>Subject: [Asterisk-Users] Asterisk capacity
>Reply-To: asterisk-users at lists.digium.com
>
>Hi,
>I am really interested in the true capacity that Asterisk can handle.
>What is the maximum number of users that can be handled by Asterisk on a
>standard 2.4G P4 IBM server or similar? Anyone has a clue?
>
>Cheer,
>Vincent
>
>_________________________________________________________________
>ninemsn Premium transforms your e-mail with colours, photos and animated
>text. Click here http://ninemsn.com.au/premium/landing.asp
>
>
>--__--__--
>
>Message: 9
>Date: Wed, 15 Oct 2003 07:53:13 +0100
>From: WipeOut <wipe_out at lycos.co.uk>
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:
> no sound after 5sec
>Reply-To: asterisk-users at lists.digium.com
>
>Steven J. Sobol wrote:
>
>
>
>>On Wed, 15 Oct 2003, Jon Pounder wrote:
>>
>>
>>Nothing works. Call transfer and call waiting, in particular. (Well,
>>almost nothing; vm notification does work)
>>
>>
>>
>>
>Call transfer and call waiting do work, although the call waiting is a
>little loud and anoyoing.. :)
>
>
>
>>There is no place to plug in a headset, and since I do a fair amount of
>>tech support and longish conference calls, that's a big deal for me.
>>
>>
>>
>A very valid point, but I also have a Snom200 and I have never been able
>to get the headset (audiojacks on the back) to work.. and this has been
>through about 10 firmware versions..
>
>So which is worse, No headset jacks or headset jacks that never work?? :)
>
>The only real issue I have with the GS phones is that the ethernet ports
>are 10Mbps ports.. So this is a problem because most people use 100Mbps
>networks now.. So I would have to run seperate cables to the phone and
>PC which kills a major cost saving for going VoIP..
>
>Later..
>
>
>--__--__--
>
>Message: 10
>Date: Wed, 15 Oct 2003 08:56:42 +0200
>Subject: Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
>From: Paul Cheng <asterisk at klarium.com>
>To: asterisk-users at lists.digium.com
>Reply-To: asterisk-users at lists.digium.com
>
>Our experience with the Budget Tones 101have been poor as well. The
>devices seem to die after a day or two (even with the new firmware) and
>then need to be rebooted. On occasion, the device needs to be literally
>unplugged and plugged back in as even the reset doesn't work.
>
>There are some nice features, but we have all but given up on them for
>a production environment. Relative to the Cisco ATAs and other devices
>we are using, the price/performance ratio is not there, particularly
>from a support cost perspective. If they get the thing to be more
>stable then we will reconsider them.
>
>On Wednesday, October 15, 2003, at 07:00 AM, rnc Info Lists wrote:
>
>
>
>>Do you have a 100 or 101? You have indicated different models in your
>>postings. Were you able to get Call Transfer and Call Waiting working
>>with your Asterisk system and other phones? Which version of the
>>Grandstream firmware do you use? There most recent on their website
>>this
>>weekend was at least 2 version numbers higher than what came on my
>>phone
>>in August. Think that they are making improvements rather frequently.
>>
>>
>>Robert
>>
>>
>>
>>
>>>On Wed, 15 Oct 2003, Jon Pounder wrote:
>>>
>>>
>>>
>>>>>The Grandstream 101 I'm using is a piece of junk but I don't have
>>>>>the
>>>>>
>>>>>
>>>>same
>>>>
>>>>
>>>>>problem with it.
>>>>>
>>>>>
>>>>What don't you like about the grandstream ? (I am not looking to
>>>>flame
>>>>you,
>>>>but was considering buying and if there are problems would rather
>>>>find
>>>>out
>>>>beforehand)
>>>>
>>>>
>>>Nothing works. Call transfer and call waiting, in particular. (Well,
>>>almost nothing; vm notification does work)
>>>
>>>There is no place to plug in a headset, and since I do a fair amount
>>>of
>>>tech support and longish conference calls, that's a big deal for me.
>>>
>>>However, keep in mind that I have an old, no-longer-manufacturered
>>>model
>>>(the Budgetone 100). Don't take my frustration with my outdated phone
>>>as
>>>a sign that you should dismiss Grandstream out of hand - I just don't
>>>like
>>>my 100.
>>>
>>>--
>>>JustThe.net Internet & Multimedia Services
>>>22674 Motnocab Road * Apple Valley, CA 92307-1950
>>>Steve Sobol, Proprietor
>>>888.480.4NET (4638) * 248.724.4NET * sjsobol at JustThe.net
>>>
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>
>
>
>--__--__--
>
>Message: 11
>Date: Wed, 15 Oct 2003 17:01:07 +1000
>From: Anthony Wood <woody+asterisk at switchonline.com.au>
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] e100p in Australia
>Reply-To: asterisk-users at lists.digium.com
>
>On Wed, Oct 15, 2003 at 12:50:33PM -0500, Stephen Dredge wrote:
>
>
>>I've seen this question asked before but haven't seen a definative answer.
>>Does the e100p work in australia? Did any one who was asking the question
>>
>>
>
>You need written permission from your Telco to use non-approved hardware,
>I think US $595 + AU $10,000 fine is more than US$1495 :-)
>
>
>
>>before bite the bullet and get one? I can get a te410 if i really have to but
>>would prefer to stay with the cheaper option.
>>
>>
>
>NetJet-S apparently works with ISDN BRI (E.g. onramp 2) and Asterisk, but there
>are some echo issues for the local end of the call.
>
>There was a Melbourne guy using one, I'm not sure how he's gone in the last 4 months:
>
>
>
>>-----Original Message-----
>>From: asterisk-users-admin at lists.digium.com
>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Mark
>>McKibbin
>>Sent: Friday, 13 June 2003 1:01 p.m.
>>To: asterisk-users at lists.digium.com
>>Subject: [Asterisk-Users] E1 cards
>>
>>
>>We are not having any luck with the E100p card here in Australia, it
>>will work with a crossover cable to another device but will not talk to
>>our Telco Telstra who probably have a weird implementation of an E1.
>>
>>Any suggestions on a replacement?
>>
>>Regards
>>
>>Mark McKibbin
>>DCS Internet
>>64 Queen St
>>Warragul
>>Victoria 3820
>>Australia
>>www.dcsi.net.au
>>mark at team.dcsi.net.au
>>Ph. 1300 665575
>>Fx. 1300 556595
>>
>>
hello every body,
First i'm sory for my bad english. i'm a newbie with voip and i want to
build a litle voip sytem for my house :
2 voip phone (softphone or not?)
1 pstn gateway (using asterisk)
I need some help to chose a pstn card (whish works with the france pstn)
and a voip phone.
I need so some help to configure asterisk.
thanks for all.
Nicolas
More information about the asterisk-users
mailing list