[Asterisk-Users] Out of 5 Grandstream BudgeTone 101 THREE are
Michael Crown
mike at thevoipconnection.com
Tue Jan 18 09:06:55 MST 2005
Ronald,
Grandstream products have a one year warrantee. If you don't have any luck
with Pulver, contact us and we can probably get your phones exchanged.
Please don't assume that your experience with Grandstream is typical. We
sell a lot of these phones and the overwhelming majority of the purchasers
are very happy with their units. The quality has improved tremendously over
the last year, and I think it might be possible that Pulver has a stock of
older units that were not as good as the ones currently shipping. We
certainly don't see that kind of failure rate as being typical.
Michael Crown
Managing Partner
The VoIP Connection
vox: 321.989.6728 ext. 611
fax: 321.989.0284
email:mike at thevoipconnection.com
-----Original Message-----
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Today's Topics:
1. Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from
Pulverstore) (Ronald Wiplinger)
2. Re: Dial Plan Agents (1 of 2) agent-dialplan.conf (Michael Loftis)
3. Number of Calls per Proxy on Cisco 7960G? (Glenn Powers)
4. RE: Is anybody using an IAXy? (Nabeel Jafferali)
5. RE: Number of Calls per Proxy on Cisco 7960G? (Nabeel Jafferali)
6. Re: Auto Protocol (depending upon registration.... (Freddi Hansen)
7. Re: RE: Issue compiling zaptel on FC 3 kernel 2.6.10-1.737
(Eric Bishop)
8. Re: Out of 5 Grandstream BudgeTone 101 THREE are defect !!!
(from Pulverstore) (el Flynn)
9. fax over tdm400p (Sergio)
10. Best Grandstream firmware to use? (Paul Fielding)
11. RE: Best Grandstream firmware to use? (David Norton)
12. Re: Best Grandstream firmware to use? (Yair Hakak)
13. Re: Wait(n) -v- Background(silence/n) ? (Tony Mountifield)
14. Re: France has their (first?) SIP carrier with "unlimited"
calls for 6eu/mo (Remco Barende)
----------------------------------------------------------------------
Message: 1
Date: Tue, 18 Jan 2005 15:46:24 +0800
From: Ronald Wiplinger <ronald at elmit.com>
Subject: [Asterisk-Users] Out of 5 Grandstream BudgeTone 101 THREE are
defect !!! (from Pulverstore)
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Cc: Diana Caporale <dcaporale at pulver.com>,
"@pulverinnovations.com"@lists.digium.com
Message-ID: <41ECBED0.9000608 at elmit.com>
Content-Type: text/plain; charset=us-ascii; format=flowed
I bought three plus two Grandstream BudgeTone 101 phones.
The shipping cost more than the phone itself from Pulver store.
The first shipping had one phone defect. Nothing on the display. (Can
happen!)
The second shipment had one phone with a defect display, but it still
worked.
The second phone's handset was defect too (microphone did not work).
Changing the handset from this one to the other one, "repaired" one of the
three defect phone sets.
NOW the next question. What is with the warranty?
Jeff Pulver & his team is silent!
In case I do not get the info for the warranty replacements I will cancel
the credit card for the purchase!
In the meantime I suggest to all of you:
1. Don't buy Grandstream!
2. (xxxx) !
Ronald
very angry Pulver customer!!!
------------------------------
Message: 2
Date: Tue, 18 Jan 2005 00:52:25 -0700
From: Michael Loftis <mloftis at wgops.com>
Subject: Re: [Asterisk-Users] Dial Plan Agents (1 of 2)
agent-dialplan.conf
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <151C14CC461514780BCDA19B@[10.1.2.77]>
Content-Type: text/plain; charset=us-ascii; format=flowed
Oh i forgot to mention....
I have found a limitation....calls going through the queue system can NOT be
parked properly. More precisely with my stdexten macro and/or the agent
logic stuff the calls can NOT be rang-back to the original extension. They
end up (in my example) in from-sip,s,1 which equates to default,s,1 but they
have ALL the internal extensions and dial plan.
Why? Heck if I know. Somehow the C code loses track of who I'm dialling
and in 1.0.1 chan_park can't find the origianl extension in the event of a
timeout. Yup you could code aroudn this in the dial plan logic by leaving
some sort of hint, but I don't get why it's missing.
Also don't put a /n at the end of the Dial(Local...) stuff in the
AgentCallBack macro, it will cause zombies, lots of them, and weird
behaviour of 7940 and 7960 SIP phones. Why? Again, don't know. I'm simply
saying 'here there be dragons' and not going in there :)
It DOES work and VERY reliably in practice, just there are the above
caveats. Sorry I forgot them in the original message.
------------------------------
Message: 3
Date: Tue, 18 Jan 2005 03:00:12 -0500
From: Glenn Powers <glenn at net127.com>
Subject: [Asterisk-Users] Number of Calls per Proxy on Cisco 7960G?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41ECC20C.5040608 at net127.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Does anyone know how many simultaneous calls per proxy I can
recieve/place on a Cisco 7960G?
thanks,
glenn
------------------------------
Message: 4
Date: Tue, 18 Jan 2005 03:02:13 -0500
From: "Nabeel Jafferali" <nabeel at jafferali.net>
Subject: RE: [Asterisk-Users] Is anybody using an IAXy?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<469C470168B32B49BB837513496593EB01CF447E at NTXBEUS01.exchange.xchg>
Content-Type: text/plain; charset="US-ASCII"
> > user: aaabbb
> > pass: cccddd
> > register
> >
> > iax.conf:
> > =========
> > [623] ; IAXy
iax.conf should read:
[aaabbb]
username=aaabbb
...
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeel<at>jafferali.net
------------------------------
Message: 5
Date: Tue, 18 Jan 2005 03:02:32 -0500
From: "Nabeel Jafferali" <nabeel at jafferali.net>
Subject: RE: [Asterisk-Users] Number of Calls per Proxy on Cisco
7960G?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<469C470168B32B49BB837513496593EB01CF447F at NTXBEUS01.exchange.xchg>
Content-Type: text/plain; charset="US-ASCII"
> Does anyone know how many simultaneous calls per proxy I can
> recieve/place on a Cisco 7960G?
Two.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeel<at>jafferali.net
------------------------------
Message: 6
Date: Tue, 18 Jan 2005 09:18:39 +0100
From: Freddi Hansen <fh at danovation.dk>
Subject: [Asterisk-Users] Re: Auto Protocol (depending upon
registration....
To: Asterisk-Users at lists.digium.com
Message-ID: <41ECC65F.9020708 at danovation.dk>
Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Subject:
> [Asterisk-Users] Auto Protocol (depending upon registration....
> From:
> "Gary" <gary at ausmail.com>
> Date:
> Tue, 18 Jan 2005 17:06:08 +1000
>
> To:
> "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com>
>
>
>Hi folks,
>
>I'm sure I had this in a previous life
>
>Basically the ability to dial with autoselection of either IAX2 or SIP
>depending upon the registration of the endpoint.
>
>Ok, I have probably missed it in the wiki as well.....
>
>hints ?
>
>Gary
>
>
Use ChanIsAvail(SIP/mylogin&IAX2/mylogin), and then Dial(${AVAILCHAN})
eventually use a macro.
Freddi
>
>
>
------------------------------
Message: 7
Date: Tue, 18 Jan 2005 19:16:12 +1100
From: Eric Bishop <asterisk.eric at gmail.com>
Subject: Re: [Asterisk-Users] RE: Issue compiling zaptel on FC 3
kernel 2.6.10-1.737
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4acda1b4050118001614e28d53 at mail.gmail.com>
Content-Type: text/plain; charset=US-ASCII
I too had the exact same issue today with FC2 and both stock and
vanilla 2.6.9 kernels... still remains unresolved. I think it could be
a broken CVS -stable......
On Mon, 17 Jan 2005 13:29:58 -0500, David Petruzzella
<dpetruzz at smartcarpet.com> wrote:
>
>
>
> I am unable to compile the zaptel drivers on the latest kernel for fc 3, I
> get the following errors which are listed below if anyone has any
> suggestions on how I can solve this issue aside from trying a different
> distro, please don't hesitate to offer. Thanks in advance.
>
>
>
> [root at asterisk-test2 zaptel]# make linux26
>
> make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel modules
>
> make[1]: Entering directory
> `/usr/src/redhat/BUILD/kernel-2.6.10/linux-2.6.10'
>
> CC [M] /usr/src/zaptel/wcfxs.o
>
> /usr/src/zaptel/wcfxs.c: In function `__check_battdebounce':
>
> /usr/src/zaptel/wcfxs.c:2193: error: `battdebounce' undeclared (first use
in
> this function)
>
> /usr/src/zaptel/wcfxs.c:2193: error: (Each undeclared identifier is
reported
> only once
>
> /usr/src/zaptel/wcfxs.c:2193: error: for each function it appears in.)
>
> /usr/src/zaptel/wcfxs.c: At top level:
>
> /usr/src/zaptel/wcfxs.c:2193: error: `battdebounce' undeclared here (not
in
> a function)
>
> /usr/src/zaptel/wcfxs.c:2193: error: initializer element is not constant
>
> /usr/src/zaptel/wcfxs.c:2193: error: (near initialization for
> `__param_battdebounce.arg')
>
> /usr/src/zaptel/wcfxs.c: In function `__check_battthresh':
>
> /usr/src/zaptel/wcfxs.c:2194: error: `battthresh' undeclared (first use in
> this function)
>
> /usr/src/zaptel/wcfxs.c: At top level:
>
> /usr/src/zaptel/wcfxs.c:2194: error: `battthresh' undeclared here (not in
a
> function)
>
> /usr/src/zaptel/wcfxs.c:2194: error: initializer element is not constant
>
> /usr/src/zaptel/wcfxs.c:2194: error: (near initialization for
> `__param_battthresh.arg')
>
> /usr/src/zaptel/wcfxs.c: In function `__check_alawoverride':
>
> /usr/src/zaptel/wcfxs.c:2195: error: `alawoverride' undeclared (first use
in
> this function)
>
> /usr/src/zaptel/wcfxs.c: At top level:
>
> /usr/src/zaptel/wcfxs.c:2195: error: `alawoverride' undeclared here (not
in
> a function)
>
> /usr/src/zaptel/wcfxs.c:2195: error: initializer element is not constant
>
> /usr/src/zaptel/wcfxs.c:2195: error: (near initialization for
> `__param_alawoverride.arg')
>
> /usr/src/zaptel/wcfxs.c:2193: error: __param_battdebounce causes a section
> type conflict
>
> /usr/src/zaptel/wcfxs.c:2194: error: __param_battthresh causes a section
> type conflict
>
> /usr/src/zaptel/wcfxs.c:2195: error: __param_alawoverride causes a section
> type conflict
>
> make[2]: *** [/usr/src/zaptel/wcfxs.o] Error 1
>
> make[1]: *** [_module_/usr/src/zaptel] Error 2
>
> make[1]: Leaving directory
> `/usr/src/redhat/BUILD/kernel-2.6.10/linux-2.6.10'
>
> make: *** [linux26] Error 2
>
> [root at asterisk-test2 zaptel]#
>
>
>
>
>
> David Petruzzella
>
> IT Department
>
> Smart Carpet Incorporated
>
> 1646 Beaver Dam Road
>
> PT. Pleasant, NJ 08742
>
> 732-899-9840
>
> www.smartcarpet.com
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
------------------------------
Message: 8
Date: Tue, 18 Jan 2005 16:21:04 +0800
From: el Flynn <el_flynn at lanvik-icu.com>
Subject: Re: [Asterisk-Users] Out of 5 Grandstream BudgeTone 101 THREE
are defect !!! (from Pulverstore)
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41ECC6F0.4030508 at lanvik-icu.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Ronald Wiplinger wrote:
> I bought three plus two Grandstream BudgeTone 101 phones.
> The shipping cost more than the phone itself from Pulver store.
>
> The first shipping had one phone defect. Nothing on the display. (Can
> happen!)
>
> The second shipment had one phone with a defect display, but it still
> worked.
> The second phone's handset was defect too (microphone did not work).
> Changing the handset from this one to the other one, "repaired" one of
> the three defect phone sets.
>
>
> NOW the next question. What is with the warranty?
>
> Jeff Pulver & his team is silent!
>
> In case I do not get the info for the warranty replacements I will
> cancel the credit card for the purchase!
>
> In the meantime I suggest to all of you:
> 1. Don't buy Grandstream!
> 2. (xxxx) !
>
> Ronald
> very angry Pulver customer!!!
>
Hmm... I've bought six BT-101s, although not from Pulver, but they haven't
given
me any problems as yet. Upgraded them all to firmware version 1.0.5.16 and
they
can now do supervised transfers.
Perhaps Pulver had a shipment of bad phones?
Flynn
------------------------------
Message: 9
Date: Tue, 18 Jan 2005 09:24:49 +0100
From: Sergio <mlists at c-net.it>
Subject: [Asterisk-Users] fax over tdm400p
To: asterisk-users at lists.digium.com
Message-ID: <41ECC7D1.8000007 at c-net.it>
Content-Type: text/plain; charset=us-ascii; format=flowed
I'm unable to get faxes working over tdm400p (4fxs modules)
Too many errors sending and receiving faxes with an analog fax
1) echocancel=no on the zap channels
2) ztmonitored the channel for a good/low audio volume
I'm trying to send fax between zap fxs channels. No way to get it
working right
Has someone else the same problem?
------------------------------
Message: 10
Date: Tue, 18 Jan 2005 01:34:47 -0700
From: Paul Fielding <paul.fielding at shaw.ca>
Subject: [Asterisk-Users] Best Grandstream firmware to use?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <02d801c4fd38$9330eeb0$0400a8c0 at MATHILDA>
Content-Type: text/plain; charset="iso-8859-1"
I've seen lots of stuff go around about Grandstream firmware levels (in my
case specifically the BT101/102). I'm just wondering what the currently
accepted 'best' firmware version is to use? After seeing stuff going around
about buggy firmware I want to know what I'm getting into before upping past
my current 1.0.5.11. It's relatively stable, and the last thing I want to
do is update to a flaky firmware....
Paul
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Message: 11
Date: Tue, 18 Jan 2005 10:50:30 +0200
From: "David Norton" <asterisk at tsol.co.za>
Subject: RE: [Asterisk-Users] Best Grandstream firmware to use?
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <E1Cqp4u-00033e-00 at guinness.msol.co.za>
Content-Type: text/plain; charset="us-ascii"
I've been using 1.0.5.16 for more than a week now, haven't had a single
problem, and have not had to reboot it once.
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul Fielding
Sent: Tuesday, January 18, 2005 10:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Best Grandstream firmware to use?
I've seen lots of stuff go around about Grandstream firmware levels (in my
case specifically the BT101/102). I'm just wondering what the currently
accepted 'best' firmware version is to use? After seeing stuff going around
about buggy firmware I want to know what I'm getting into before upping past
my current 1.0.5.11. It's relatively stable, and the last thing I want to
do is update to a flaky firmware....
Paul
--
This message has been scanned for viruses and
dangerous content and is believed to be clean.
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------------------------------
Message: 12
Date: Tue, 18 Jan 2005 10:55:01 +0200
From: Yair Hakak <yhakak at gmail.com>
Subject: Re: [Asterisk-Users] Best Grandstream firmware to use?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43ff339405011800552275aff at mail.gmail.com>
Content-Type: text/plain; charset=US-ASCII
i've actually had reboot issues since moving to 1.0.5.16, the phones
seem to hang more often on soft reboot and require a hard reboot
(unplugging). This is just a feeling and i can't quantify this but i
don't remember having to physically reboot the phones this often
before. I'm using one bt-101 and one bt-102.
-yair
On Tue, 18 Jan 2005 10:50:30 +0200, David Norton <asterisk at tsol.co.za>
wrote:
>
>
> I've been using 1.0.5.16 for more than a week now, haven't had a single
> problem, and have not had to reboot it once.
>
>
> ________________________________
>
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul
Fielding
> Sent: Tuesday, January 18, 2005 10:35 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Best Grandstream firmware to use?
>
>
>
>
>
> I've seen lots of stuff go around about Grandstream firmware levels (in my
> case specifically the BT101/102). I'm just wondering what the currently
> accepted 'best' firmware version is to use? After seeing stuff going
around
> about buggy firmware I want to know what I'm getting into before upping
past
> my current 1.0.5.11. It's relatively stable, and the last thing I want
to
> do is update to a flaky firmware....
>
>
>
>
>
> Paul
> --
> This message has been scanned for viruses and
> dangerous content and is believed to be clean.
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
------------------------------
Message: 13
Date: Tue, 18 Jan 2005 08:08:55 +0000 (UTC)
From: tony at softins.clara.co.uk (Tony Mountifield)
Subject: [Asterisk-Users] Re: Wait(n) -v- Background(silence/n) ?
To: asterisk-users at lists.digium.com
Message-ID: <csig6n$50r$1 at softins.clara.co.uk>
In article <1106023295.21104.10.camel at critch>,
Steven Critchfield <critch at basesys.com> wrote:
> On Tue, 2005-01-18 at 10:44 +1100, Howard Lowndes wrote:
> > Will Wait(n) still listen for DTMF input from the caller after there has
> > been a Background(some-message) prompt, or do I need to use
> > Background(silence/n) to still listen for DTMF?
>
> You don't need anything but a proper gap. You need to program the
> extensions like you do with a event loop.
>
> exten => s,1,Wait,0
> exten => s,2,Answer
> exten => s,3,DigitTimeout,5
> exten => s,4,ResponseTimeout,10
> exten => s,5,BackGround,demo-congrats
>
> ; This is a blank area that just waits to get DTMF for up to 10
> ; seconds due to the ResponseTimeout
>
> exten => t,1,Goto(somewhere-due-to-timeout)
What's the reason for having a zero-length Wait befor the Answer?
Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
------------------------------
Message: 14
Date: Tue, 18 Jan 2005 10:03:49 +0100 (CET)
From: Remco Barende <asterisk at barendse.to>
Subject: Re: [Asterisk-Users] France has their (first?) SIP carrier
with "unlimited" calls for 6eu/mo
To: Asterisk Users List <asterisk-users at lists.digium.com>
Message-ID: <Pine.LNX.4.61.0501181003290.16971 at raveon.vaag.nu>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
On Mon, 17 Jan 2005, Wilson Pickett wrote:
>> Can you offer any clue where I would need to look, I guess its not in
>> extensions.conf that is the problem?
>
> 1) Are you registering with proxy1?
no, from the earlier post I understood that I shouldn't register with the
proxy? I guess that means that I should setup 2 SIP entries, one for the
outgoing calls (that registers with len) and another one for the incoming
calls
(that registers with proxy)?
> 2) You'll need a user or friend entry as well as the peer - at least
> that's what I did to get it working. The peer uses len1 and the friend
> uses proxy1
I tried setting it that way, but still do not get through.
sip show does show 2 connections now
Would you mind sending me the relevant bits of your sip.conf?
Thanks!!!
------------------------------
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