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:[EMAIL PROTECTED] -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 18, 2005 4:04 AM To: [email protected] Subject: Asterisk-Users Digest, Vol 6, Issue 256 Send Asterisk-Users mailing list submissions to [email protected] 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." 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 <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore) To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Cc: Diana Caporale <[EMAIL PROTECTED]>, "@pulverinnovations.com"@lists.digium.com Message-ID: <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Dial Plan Agents (1 of 2) agent-dialplan.conf To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Number of Calls per Proxy on Cisco 7960G? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Is anybody using an IAXy? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Number of Calls per Proxy on Cisco 7960G? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Re: Auto Protocol (depending upon registration.... To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii; format=flowed > > Subject: > [Asterisk-Users] Auto Protocol (depending upon registration.... > From: > "Gary" <[EMAIL PROTECTED]> > Date: > Tue, 18 Jan 2005 17:06:08 +1000 > > To: > "[email protected]" <[email protected]> > > >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 <[EMAIL PROTECTED]> 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 <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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. > > > > [EMAIL PROTECTED] 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 > > [EMAIL PROTECTED] 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 > [email protected] > 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 <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore) To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> Subject: [Asterisk-Users] fax over tdm400p To: [email protected] Message-ID: <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Best Grandstream firmware to use? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/1d3c83 e1/attachment-0001.htm ------------------------------ Message: 11 Date: Tue, 18 Jan 2005 10:50:30 +0200 From: "David Norton" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Best Grandstream firmware to use? To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/4f9fe0 67/attachment-0001.htm ------------------------------ Message: 12 Date: Tue, 18 Jan 2005 10:55:01 +0200 From: Yair Hakak <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Best Grandstream firmware to use? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 > [email protected] > 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: [EMAIL PROTECTED] (Tony Mountifield) Subject: [Asterisk-Users] Re: Wait(n) -v- Background(silence/n) ? To: [email protected] Message-ID: <[EMAIL PROTECTED]> In article <[EMAIL PROTECTED]>, Steven Critchfield <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ------------------------------ Message: 14 Date: Tue, 18 Jan 2005 10:03:49 +0100 (CET) From: Remco Barende <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] France has their (first?) SIP carrier with "unlimited" calls for 6eu/mo To: Asterisk Users List <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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!!! ------------------------------ _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 6, Issue 256 ********************************************** _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
