Re: [Asterisk-Users] Problem with date time on Aastra480isincerelease 1.3
That exactly what I had to do to get it working. Very very weird... Seems like a bug in 1.3 Lee Archer wrote: Actually it worked, but only after I defaulted all the settings on the phone and let it pick the config up fresh. Anyone know if there is any headset config options to default to headset/speaker? Thanks Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Lee Archer Sent: 03 January 2006 14:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem with date time on Aastra480isincerelease 1.3 Still no joy, if I set my phone to a different time zone then reboot it isn't being updated to use London. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Pete Barnwell Sent: 03 January 2006 14:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem with date time on Aastra 480isincerelease 1.3 On Tue, 2006-01-03 at 14:13 +, Lee Archer wrote: Does anyone know whether there is some sort of time zone option? I've emailed Aastra who didn't come back to me. I would like to set the time zone - e.g. Britain-London, in the cfg files so I don't have to set it on 40 phones... time zone name: GB-London time zone code: GMT time zone minutes: 60 Rgds Pete ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Page application do?
Can this work with any ADSI phone? Can you send some links. The documentation is quite hard to find.. Thanks Jacques Andrew Latham wrote: I think most all of the phones have a hack to get it working. Aastra analog ADSI phones even work as I read some where... On 12/29/05, Robert La Ferla [EMAIL PROTECTED] wrote: So I can set it up to call a bunch of extensions and broadcast a message to them without the user picking up? Can I do this with Aastra phones? This would be great for announcing incoming calls. "You have a call from XXX . Press 1 to pickup Press 2 to send them to voicemail." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stay away from Grandstream!
My only experience is with their Budgetone 102. You basically get you pay for. I have since purchased a pair of Aastra 480i. Much much better. I am going to put the Budgetone on ebay, no point dealing with all the hassle. The main issue for me was actually not sofware but rather the design of the handset. Vahan Yerkanian wrote: Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their final firmware development costs by providing free testing/QA. I blame their sales management for pushing developers to release without proper testing. GXP2000 is much more buggy echo-can wise than the earlier models. For now, I'm back to more expensive equipment. We're not that rich to pay twice. HTH, Vahan Avi Miller wrote: Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in Sydney, I'm probably going to use a mix of Polycom/Grandstream and softphones. cYa, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with date time on Aastra 480i since release 1.3
Thanks Robert. I tried of course with time server disabled: 0 too. Is it working for you? Which time server are you using, an external one? Robert La Ferla wrote: Jacques Leisy wrote: Since the release 1.3 the 480i displays the wrong date and time. Something in 1947 ! I have followed the settings in the aastra.cfg. time server disabled: 1 time server1: 192.168.0.10 time server2: 192.168.0.11 # time server3: 128.121.51.132 time format: 1 date format: 0 My servers are running the proper time server. Same problem when I connect to the roku time server. Am I missing one entry? To enable the time server, you need: time server disabled: 0 1 means disabled 0 means enabled ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with date time on Aastra 480i since release 1.3
Since the release 1.3 the 480i displays the wrong date and time. Something in 1947 ! I have followed the settings in the aastra.cfg. time server disabled: 1 time server1: 192.168.0.10 time server2: 192.168.0.11 # time server3: 128.121.51.132 time format: 1 date format: 0 My servers are running the proper time server. Same problem when I connect to the roku time server. Am I missing one entry? Thanks Jacques ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard TDM2400P: comments
Can you define a LOT of pots line? Have you considered a channel bank. Here I'm running an ADTRAN 750. It's painless. You just need 1 T1 interface card for 24 lines. Jacques yusuf wrote: Hi all, we have the need for alot of plain analog lines. We thinking of buying the new Wildcard TDM2400P. Does anybody have any comments with using this card, with any version of Asterisk, (maybe ill make this one Asterisk 1.2.x). I have had some stabilty issues using the 4 TDM400P. What about this new TDM2400P??? thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Compact PCI platform
Anybody running * on a compact PCI platform? I got a few CPCI boards on eBay including a T1 Natural Microsystems AG4000? Any hope to ever get * running on that platform? Linux Suse 9.0 is running fine Thanks Jacques
[Asterisk-Users] Fuse for Adtran 750 PSU
Sorry for a very stupid question, but I cannot find a supplier anywhere. Where can I buy the 3 Amps GMT fuses for the Adtran's PSU. Car fuse don't seems to fit. What is GTM the abbreviation of Thanks Jacques
[Asterisk-Users] Is there a BIG difference between a softphone like X-Lite and a hard VOIP phone
Before I spend an extra $100 dollars for a "hard phone", I would like to know if I can expect a significant delay reduction and quality improvement over a product like X-Lite. Of course it will depend so here are a few pointers on my config: Workstation -- - 1.4Ghz AMD Athlon - Windows XP Professional - 802.11g connection to a 100Mhz switched network I tried with different codecs (uLaw, GSM) and could not detect any significant difference. Server -- - Intel Celeron 1.15Mhz - Suse Linux 9.0 - Kernel 2.4.21 - Asterisk CVS-02/03/04 (still trying to find out the command to check for the version) A few things I need to try - impact of wired vs wireless - impact of processor upgrade on the server (I'm going to install a standalone server with a dual pentium and also try the CompactPCI motorola server I bought last week end !!!) Thanks Jacques
RE: [Asterisk-Users] Calling SIP
Thanks Eric. I'll configure my system for IAXTEL today and try it Have a great week end -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Saturday, February 21, 2004 8:11 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Calling SIP Thanks for the reminder, I forgot to change my web page and .sig when I moved. You can access my public demo services via 1) IAXTel 1-700-923-3656 x2101 2) PSTN 228-467-9866 x2101 or 3) (the recommended way) Dial(IAX2/[EMAIL PROTECTED]/2101) Not all the services are working, the call back demo is not available, and the weather report is missing some info since weather.com reworked their homepage. On Sat, 2004-02-21 at 18:19, Jacques Leisy wrote: Eric, I checked your page . Very interesting, thanks! I tried to call the number indicated ...IAXTel number 700-923-3645. My PSTN number is 850-484-4535. The extension for System Services is 2101... But I got a disconnected message. After that I called the number listed at the bottom of this email (850-484-4545) expecting a system prompt but a women answered the phone. Sorry for the inconvenience. If I want to try your scripts without bothering anyone, what is the proper # Thanks Jacques -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, February 09, 2004 2:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Calling SIP That's just the way Asterisk's dial command works. On Mon, 2004-02-09 at 13:16, Tim Sailer wrote: I've looked, poked, and hoped, but I can't seem to make * understand the difference between a SIP channel being busy or not being there. Both come up as 'busy'. I would expect the unregistered SIP to be seen as unavailable. Am I just missing something obvious, again? Tim -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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 ___ 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 -- Eric Wieling [EMAIL PROTECTED] BTEL Consulting ___ 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 ___ 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
[Asterisk-Users] What is the best way to debug the DTMF tones on a Zap interface
I've started the integration of * with my PBX and I need to get a good understanding on the tones sent by it on the AA/VM port. What is the best way to do that. I could create an AGI but I'm afraid to loose some information Thanks Jacques
RE: [Asterisk-Users] Calling SIP
Eric, I checked your page . Very interesting, thanks! I tried to call the number indicated ...IAXTel number 700-923-3645. My PSTN number is 850-484-4535. The extension for System Services is 2101... But I got a disconnected message. After that I called the number listed at the bottom of this email (850-484-4545) expecting a system prompt but a women answered the phone. Sorry for the inconvenience. If I want to try your scripts without bothering anyone, what is the proper # Thanks Jacques -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, February 09, 2004 2:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Calling SIP That's just the way Asterisk's dial command works. On Mon, 2004-02-09 at 13:16, Tim Sailer wrote: I've looked, poked, and hoped, but I can't seem to make * understand the difference between a SIP channel being busy or not being there. Both come up as 'busy'. I would expect the unregistered SIP to be seen as unavailable. Am I just missing something obvious, again? Tim -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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 ___ 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
RE: [Asterisk-Users] Digium connectivity issue?
They moved on Friday: Hello. We will be moving Friday, February 13th to a larger facility. Therefore, we will be closed this Friday and unable to receive faxes, ship product, or recieve phone calls. Thanks in advance for your understanding. Here is our new Address: The Atrium Building Ste. 100 150 West Park Loop Huntsville, AL 35806 Seems to be working now though -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepakumar JV Sent: Saturday, February 14, 2004 7:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium connectivity issue? I am having same problem and i was never successful in connecting to digium.com or asterisk.org or asteriskpbx.org for last three days. Deepak - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Sent: Friday, February 13, 2004 01:54 PM Subject: [Asterisk-Users] Digium connectivity issue? Are others seeing hugh delays and/or lack of connectivity to Digium? Rich ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface
Are you dialing out to the public network? I thinking there is a prefix you can dial out to hide your number. Is it *67? Jacques -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mickey Binder Sent: Friday, February 13, 2004 6:14 AM To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ 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 ___ 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
RE: [Asterisk-Users] Asterisk Panasonic KXTD - Vonage
I was hoping to be able to avoid using the V1005 altogether. And have vonage call the asterisk server. I imagine it would be better quality and identification. I have it working with the X101P Jacques -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Youness El Andaloussi Sent: Sunday, February 08, 2004 2:45 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Panasonic KXTD - Vonage 2nd question (should I use a separate email?) I have Vonage service. Is it possible to end the call directly in my asterisk system rather than in the Motorola V1005? with any compatible FXO card, like digium X100P Youness ___ 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 ___ 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
[Asterisk-Users] Asterisk Panasonic KXTD - Vonage
Has anybody interfaced an asterisk system with a Panasonic KXTD switch? I have invested in a few digital phones for my house and would hate to throw them away. I'm thinking about interfacing with the KXTD using their voice mail integration. Only issue is that it is quite difficult to find any information about their protocol. Is the only solution to use their inband DTMVF signaling with an FXO card? 2nd question (should I use a separate email?) I have Vonage service. Is it possible to end the call directly in my asterisk system rather than in the Motorola V1005? Thanks Jacques
