RE: [Asterisk-Users] OT: AudioCodes MP124-C/FSX/AC/SIP
FXS or FXO? Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: William Piper [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 24, 2006 6:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OT: AudioCodes MP124-C/FSX/AC/SIP Sorry to hijack your thread. Reading these posts made me grab my old MP-104 try again to get it working with asterisk. I bought it a while ago off eBay never could get it to register. Does anyone have an example ini file for the MP-1XX that I could look at figure out what I am configuring wrong on this box? Thanks, bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, May 24, 2006 6:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: AudioCodes MP124-C/FSX/AC/SIP From my copy of the manual MP-1xx SIP User's Manual 36 Document #: LTRT-65404 You can use the 'Reset' button to restore the MP-1xx networking parameters to their factory default values (described in Table 4-1) and to reset the username and password. Note that the MP-1xx returns to the software version burned in flash. This process also restores the MP-1xx parameters to their factory settings, therefore you must load your previously backed- up ini file, or the default ini file (received with the software kit) to set them to their correct values. To restore networking parameters to their initial state, take these 6 steps: 1. Disconnect the MP-1xx from the power and network cables. MP-1xx SIP User's Manual 4. Getting Started 2. Reconnect the power cable; the gateway is powered up. After approximately 45 seconds the Ready LED turns to green and the Control LED blinks for about 3 seconds. 3. While the Control LED is blinking, press shortly on the reset button (located on the left side of the front panel); the gateway resets a second time and is restored with factory default parameters (username: Admin, password: Admin). 4. Reconnect the network cable. 5. Assigning the MP-1xx IP address (refer to Section 4.1 on page 35). 6. Load your previously backed-up ini file, or the default ini file (received with the software kit). To load the ini file via the Embedded Web Server, refer to Section 5.9.2.1 on page 120. On May 24, 2006, at 3:00 PM, Erick Perez wrote: Just a question, has anyone knows how to blank or factory reset an AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit). I purchased them second-handed with no manuals (thank god for the internet!!) but i guess the pdf manual I have does not have the section of factory-reset. Also, any sucess stories with: AudioCodes MP124-C/FSX/AC/SIP ---Asterisk---internet---Vonage setups? Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ ___ --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] Now that Nufone is dead...
Nufone is not dead. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Carlos Chavez [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 23, 2006 10:49 AM To: Asterisk Subject: [Asterisk-Users] Now that Nufone is dead... Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] Snom firmwares suck
I can confirm that the issue with the display is usually a hardware problem, not firmware. If the phone has this issue, it may be related to the other issues (locking up, etc.). I suggest you return the phones that have this problem for warrantee exchange. As far as the firmware goes, the production versions generally have a few minor issues but are pretty sound overall (relative to other vendors). -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Dr. Michael J. Chudobiak [mailto:[EMAIL PROTECTED] Sent: Friday, May 19, 2006 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Snom firmwares suck Remco Barende wrote: Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone Remco, I have a dozen Snom 360s. One had a defective LCD that would become garbled after time. Snom support quickly confirmed that this was a known issue, and my vendor (voipsupply) quickly sent a replacement. I've never seen any lockups or reboots. I reboot the phones each night at midnight, just to be safe - try doing that to see if it reduces problems. I've posted a sample perl cron script at http://www.voip-info.org/wiki/view/Asterisk+phone+snom to show how. I use shielded ethernet cables (STP) everywhere too. Try that - good grounding may be beneficial. It can't hurt, anyway. Snom support is pretty responsive. Try emailing [EMAIL PROTECTED]; they have fixed some issues for me (for example, the clock was showing the wrong time due to daylight savings time problems). Try using a Grandstream GXP-2000 phone, and you'll see why people like the Snoms :-) Hope this helps - let us know if anything makes a difference! - Mike ___ --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] WiFi VoIP Handsets..
According to all of my sources, the UIP1868 has been discontinued. Kind of a shame, it was a neat product. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Colin MacMillan [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 17, 2006 10:09 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] WiFi VoIP Handsets.. I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months.Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one ...? On 5/17/06, Andrew Latham [EMAIL PROTECTED] wrote: CoryThe 480i-CT does not state DECT to my knowlege as the EU DECT standarduses reseved frequency space in the US.I have heard rumblings abouta US DECT standard, would this be the DECT you are refering to and if so could you provide a link to information on compatablity.AndrewOn 5/16/06, Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --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-Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)[EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!Hind sight is most always 20/20 or better.---___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] grandstream GXV-3000
Marek,
We have tested that that scenario and it works fine with the dev version of
Asterisk. -Mike
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: marek cervenka [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 09, 2006 8:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] grandstream GXV-3000
hi,
do you someone test this http://www.grandstream.com/y-gxv3000.htm?
video works? (it's have H264 video codec)
i want this topology
gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000
---
Marek Cervenka
LCNA - http://lcna.slu.cz
===
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RE: [Asterisk-Users] Tool for Polycom configurations
Hi Bruce, We've written software to do this as a service for our customers. I can't give you the program, but we'd be willing to program your phones for you. Contact me off list. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Bruce Reeves [mailto:[EMAIL PROTECTED] Sent: Thursday, May 04, 2006 2:45 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Tool for Polycom configurations I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files?-- Bruce Nortex Networks ___ --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] Grandstream Budgetone and Mac mini?
The switch in the Budgetone is 10Base-T. If the PC NIC cannot auto-detect or otherwise handle that, it will be a problem. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Jerry Jones [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 18, 2006 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream Budgetone and Mac mini? Are you seeing link on either end? Not sure if the GS shows or not. On the Mac, open a terminal window and type ifconfig to see if the port is active - ie has link it should have a line similar to this if so media: autoselect (100baseTX full-duplex) status: active If this is correct then you have something else wrong On Apr 18, 2006, at 8:22 AM, Rusty Dekema wrote: The PC port on a BT-102 should work with any computer that has an Ethernet card. Have you tried these phones with other computers than the Mac Minis you mention? It shouldn't make any difference whether the computer is a Mac, PC or anything else. Perhaps something is wrong with the BT-102s you have. -Rusty On 4/18/06, Dmitry Ivanov [EMAIL PROTECTED] wrote: Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --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 ___ --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] Best ATA for general residential deployment??
Not true. There are hundreds of thousands of Grandstream adapters in use around the world. Grandstream support is not perfect, but it is as good or better better than most vendors, including Linksys/Sipura.The Grandstreams do currently have a bug with header compression right now that causes problems for some PPPoE setups, but it's getting fixed. The newest firmware is very stable overall. We work with almost every device and they all have some issues. If you consider the vast variety of different equipment that these things have to be interoperable with you can begin to appreciate how challenging it is to make them work properly. Given the dynamic nature of the environment, there will always be a certain number of situations where a given product doesn't perform. I stand by my original assertion. The Linksys line of products are also excellent, but they are considerably more expensive for the same functionality. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, April 09, 2006 9:12 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Best ATA for general residential deployment?? Grandstreams are totally useless, I had to switch all my phones to Linksys. Grandstream will not even support you and their router side do not work for the 486 or 496. -- Original message -- From: "Andre Rodrigues (Cheyenne)" [EMAIL PROTECTED] I have more than 20 ATA 386. They can not work for more than one day without a local and "hard reboot". Do no buy these ata please!!! Regards Amr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of The VoIP Connection Sent: quarta-feira, 22 de Fevereiro de 2006 23:11 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Best ATA for general residential deployment?? Absolutely. HT-486 is my pick for best all-around unit based on ease-of-use, value, performance and reliability. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best ATA for general residential deployment?? On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote: On 2/22/06, Matt <[EMAIL PROTECTED]>wrote: Yes.. there are provisioning tools that you have to get. Unfortunately it's this catch 22 loop. You have to prove that youcan offer 200+ ATAs to customers, or you can't get the tools, butyet, you don't really want to offer those ATAs to the customer'swithout having the tools. This sounds like yet another reason to avoid purchasing Sipuraequipment and supporting Sipura in any way. I don't know about you guys, but I have better things to do than screw around with asinine vendor policies that make it more difficult than necessary to get things done. True, but it's kind of a "pick your poison" situation in my opinion. Ht-486 anyone? ___ --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 ___ --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] Asterisk with Vonage
Actually, they do have a bring your own device program. It's called Business Plus. Works great with Asterisk. http://www.thevoipconnection.com/store/catalog/product_16220_Vonage_Business_Plus.html Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Steve Jones [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 29, 2006 8:59 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk with Vonage I know Vonage doesnt officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! ___ --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] Any Polycom dealer willing to help?
If you purchased your phones from an authorizedreseller they shouId be able to provide this. Ican help you. Please contact me off list. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Eric Bishop [mailto:[EMAIL PROTECTED] Sent: Monday, March 27, 2006 3:58 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Any Polycom dealer willing to help? Hi All,We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM 3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.htmlCan someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only old versions.Are there any kind Polycom authorized dealers who can help me?-- Eric ___ --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] An FXO version of IAXy?
We are in final testing and will shortly begin shipping IAX CPE devices with the following configurations: 1 FXS 2 FXS 1 FXS, 1FXO These devices have an integrated gateway router with IPSec VPN, QoS, and more. Codec support includes G.711, G.723.1, G726, and G 729a/b. Please contact me off-list for pricing and details. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Dr. Michael J. Chudobiak [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 22, 2006 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] An FXO version of IAXy? D-Link has a 4 port FXO device on their site. http://www.dlink.com/products/?sec=2pid=451 Apparently it hasn't shipped yet and costs $500.00 I've been testing a AudioCodes MP104-FXO-C3S (around $1000) 4-port FXO box. It works, but the number of configuration options are staggering, complex, and inter-related, and the documentation support just aren't good enough to make installation easy. The D-link DVG-3004S is pretty much impossible to get. There is also the Mediatrix 1104 (also around $500), but it is reputed to be hard to configure (no web interface - just snmp!). Slapping a Sangoma A200 into a computer (and configuring it through Zaptel/Asterisk) is much, much simpler than trying to make the appliance gateways work, at least in my experience. - Mike ___ --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] Polycom 501 power over ethernet
I've seen a lot of IP501 and I've never seen one with a power jack. According to Polycom they all use the cable. Possibly it was an IP500? -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Monday, March 06, 2006 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet No, some IP 501's have the inline cable and some have the power jack. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sunday, March 05, 2006 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet The IP300/301 has the power jack, the IP500/501 the inline cable. PaulH On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote: Not true. Some do and some don't. Some have a place to plug a separate DC adapter, and some have the inline power, where the adapter plugs into the ethernet cable. Not sure which ones are newer, and which are older. -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Sun 3/5/2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet The IP501 does not have a power jack. You'll need one of the Polycom cables. William M Conlon wrote: My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. Notice the advanced technical jargon here With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were powered over ethernet. Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501
RE: [Asterisk-Users] Best ATA for general residential deployment??
What a load of @#$%! The 102 is obviously a ripoff of the HandyTone 486! Even the spec sheet is copied! Like HT486 only much better? I doubt it. I have it on good authority that Grandstream is taking legal and police action against these people and if there is any justice in the world they will be out of business soon. I'm all for competition, but piracy is another thing altogether.-Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Tele Cost Price Reducer [mailto:[EMAIL PROTECTED] Sent: Thursday, February 23, 2006 9:12 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Best ATA for general residential deployment?? hi , i have some options we are working with at vast deployement with no problems: www.tigernetcom.com - type 102 is a nice ATA, like GS 486 but far away better. we import directly from the producer at great prices so if anybody interested, please contact off-list. additionaly, we know another excellent producer (price of about 45$ FOB china) we would recomend off-list. Mickey On 2/23/06, Marc Rys [EMAIL PROTECTED] wrote: My $.02 is that HT486 sucks as a router.It works well as a ATA.I have 5 and have all of them are behind separate routers.The HT486 never gave me the full download speed of my cable modem and even when my PC wasn't powered up, and I wasn't talking on the HT486 my cable modem still looked like the HT486 was sending traffic non-stop.I put a Buffalo router in front of the HT486 and all is good.Cable modem doesn't look full of activity during idle time and my PC can use the cable modem to fullest potential. It's a shame too, because I bought the HT486's for the router capability.It turned out to be a waste.Marc-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of The VoIP ConnectionSent: Wednesday, February 22, 2006 5:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Best ATA for general residential deployment??Absolutely. HT-486 is my pick for best all-around unit based on ease-of-use, value, performance and reliability. -MikeMichael CrownManaging Partnerwww.thevoipconnection.com321.989.6728 ext. 611sip:[EMAIL PROTECTED] -Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED]] Sent: Wednesday, February 22, 2006 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best ATA for general residential deployment?? On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote: On 2/22/06, Matt [EMAIL PROTECTED] wrote: Yes.. there are provisioning tools that you have to get. Unfortunately it's this catch 22 loop.You have to prove that you can offer 200+ ATAs to customers, or you can't get the tools, but yet, you don't really want to offer those ATAs to the customer's without having the tools.This sounds like yet another reason to avoid purchasing Sipura equipment and supporting Sipura in any way. I don't know about you guys, but I have better things to do than screw around with asinine vendor policies that make it more difficult than necessary to get things done. True, but it's kind of a "pick your poison" situation in my opinion. Ht-486 anyone? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.375 / Virus Database: 268.0.0/266 - Release Date: 2/21/2006--No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.375 / Virus Database: 268.0.0/266 - Release Date: 2/21/2006 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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 business IP phone to use
Don't have an SPA-942 here right now, but a D-Link switch detects the SPA-941 as 10base-T/half-duplex. Just like real Cisco phones, the 942 can be powered with a wall wart but it does not come with one (extra charge). -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 5:30 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: 'mustardman29' Subject: RE: [Asterisk-Users] What business IP phone to use On Wed, 22 Feb 2006, The VoIP Connection wrote: The 941/942 are very nice phones. They are well made and so far the firmware seems very solid, but like their Cisco brethren they are a little expensive for what they offer in my opinion. If they were 25-30% cheaper I would be a lot more enthusiastic. If the 941 was priced like the 841 it would be a homerun. does the 942 have two 10meg ports or two 100meg ports? and is it poe only, or does it have the option of being powered from a wallwart without a poe injector? -Dan ___ --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] Best ATA for general residential deployment??
Absolutely. HT-486 is my pick for best all-around unit based on ease-of-use, value, performance and reliability. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best ATA for general residential deployment?? On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote: On 2/22/06, Matt [EMAIL PROTECTED] wrote: Yes.. there are provisioning tools that you have to get. Unfortunately it's this catch 22 loop. You have to prove that you can offer 200+ ATAs to customers, or you can't get the tools, but yet, you don't really want to offer those ATAs to the customer's without having the tools. This sounds like yet another reason to avoid purchasing Sipura equipment and supporting Sipura in any way. I don't know about you guys, but I have better things to do than screw around with asinine vendor policies that make it more difficult than necessary to get things done. True, but it's kind of a pick your poison situation in my opinion. Ht-486 anyone? ___ --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 business IP phone to use
I have used every phone and talk to customers using different devices all day long and I can tell you there is no single IP phone that is perfect for everyone. You will not find the answer on a newsgroup or a wiki, you need to judge for yourself. For example, while I may love the decidedly euro ergonomics of the snom, you may find it impossibly unconventional. We have lots of customers who are very happy with their GXP-2000's as well as a number who are not. It depends on how they are being used (especially LAN or WAN) as well as the firmware version and networking environment. We also have many customers who love their Polycoms and there is no doubt that they build a quality product. They aren't cheap but they don't disappoint. By the way, Polycom officially supports Asterisk through certified resellers as of October 2005. Snoms are great also but they seem to be having some trouble getting the version 5.0 firmware stable. If you can live with the features in V4.x for a while, these phones are terrific. Probably the best overall integration with Asterisk of any IP phone currently available. Aastra seems to be getting it together at last and also are worthy of consideration. I sell phones for a living and here's what I recommend: First, select a reliable and competent vendor who will work with you (shameless plug for The VoIP Connection). Talk to them and narrow the field to a sampling of the phones you think will work for your organization. Set up a test scenario that simulates the network environment you will have and learn how to set the phones up with Asterisk (and vice-versa) so that they work the way they should. Learn how to use the features well enough to teach them (if you can't explain the basic operation of the phone in 5 minutes forget it), and then put them in front of a sampling of the people who will use them every day. Pay special attention to your receptionist and office manager since they will be the ones you will hear from the most. There really is no shortcut if you want your users to be happy. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: mustardman29 [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 21, 2006 12:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What business IP phone to use I have been struggling with this issue for about a year now. There were just too many IP phones to choose from at all sorts of price points and not enough information about any of them. Now I am looking at the situation again and if anything it has gotten worse. There are even more phones and all sorts of opinions. For every person that says phone x is great there is someone else complaining about it. I ended up buying a Grandstream GXP2000 and an Aastra 9133i to test so I pretty much know what those two phones are about. Lot's of people talking about Polycom phones but they still seem to have their problems and since they don't officially support Asterisk I have my concerns. I really don't want to have to keep buying phones to find out for myself as it get's expensive real fast. Is there any unbiased comparison of various phones and features anywhere. If someone wrote a book I'd buy it but it would probably be obsolete before it was published with the rate of new IP phone introductions and firmware revisons. I hear some people praising the GXP2000 phones and I gotta wonder what they are smokin (regardless of firmware revison) so I just don't know who to believe anymore. ___ --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 business IP phone to use
There's lots to like about the GXP-2000 in terms of features for the money and Grandstream is working very hard to make the phone work well with Asterisk. The sound is on a par with more expensive phones and many people find the clean, minimalist look of the GXP-2000 appealing. The ergonomics are also very familiar for Americans. Again, personal taste factors into the mix as does budget. Some people can't just stand rubber buttons, some don't like plastic. We have been watching the Aastra line for about two years now waiting for the firmware to be ready for primetime and we are currently in the process of adding them to our catalog. They are certainly a capable group and we have also found them to be easy to deal with. The reality is that they are a little late to the game with a viable offering and they have some catching up to do, but their progress is encouraging. The 941/942 are very nice phones. They are well made and so far the firmware seems very solid, but like their Cisco brethren they are a little expensive for what they offer in my opinion. If they were 25-30% cheaper I would be a lot more enthusiastic. If the 941 was priced like the 841 it would be a homerun. Polycom,like most of the higher end manufacturers, supports the user through their channel. If you buy your phones from a cut-rate or unauthorized reseller you will not get good support. Factor it into your decision making process. And finally, you don't need to fly to a trade show to try a variety of phones. If you contact us we can set you up with a 30 day test program. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: mustardman29 [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 21, 2006 11:25 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] What business IP phone to use Thanks Michael, That sounds like good advice. I am surprised that some customers like the GXP2000. Cheap looking, cheap sounding, high failure rates. What sort of customers are we talking about within the context of business users if you don't mind me asking? Not home users. Business users in office environments. I have been gravitating towards the Aastra's because I like the features/price points the 3 flavors hit. I also really like the support I have been get from the manufacturer of the phones and firmware. I have been patiently waiting for the firmware to improve and I think it is just about there now. I do have concerns about Polycom's arms length attitude towards the end user but knowing they now sort of support Asterisk is a good thing. I can see why you would advise to find a good reseller for Polycom's. I guess I will have to fly out to a VoIP trade show somewhere where I can touch and use a bunch of different phones without having to buy them. Anyone have any opinions on the Linksys 941/942? It sounds like the firmware is ok but my main concern is always the hardware which won't really improve over time like firmware. What are the handset/speakerphone/buttons like compared to GXP2000, Aastra480, Aastra9133i, Polycom 501 etc. Any info would be greatly appreciated. -Original Message- From: The VoIP Connection [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 21, 2006 12:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] What business IP phone to use I have used every phone and talk to customers using different devices all day long and I can tell you there is no single IP phone that is perfect for everyone. You will not find the answer on a newsgroup or a wiki, you need to judge for yourself. For example, while I may love the decidedly euro ergonomics of the snom, you may find it impossibly unconventional. We have lots of customers who are very happy with their GXP-2000's as well as a number who are not. It depends on how they are being used (especially LAN or WAN) as well as the firmware version and networking environment. We also have many customers who love their Polycoms and there is no doubt that they build a quality product. They aren't cheap but they don't disappoint. By the way, Polycom officially supports Asterisk through certified resellers as of October 2005. Snoms are great also but they seem to be having some trouble getting the version 5.0 firmware stable. If you can live with the features in V4.x for a while, these phones are terrific. Probably the best overall integration with Asterisk of any IP phone currently available. Aastra seems to be getting it together at last and also are worthy of consideration. I sell phones for a living and here's what I recommend: First, select a reliable and competent vendor who will work with you
RE: [Asterisk-Users] Grandstream Budgetone mass deployment?
We do this routinely as a service for our customers. How many phones do you need to provision? Do you already have the phones? Contact me off list and I can help you out. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Dmitry Ivanov [mailto:[EMAIL PROTECTED] Sent: Monday, January 30, 2006 4:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream Budgetone mass deployment? Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working? ___ --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] Home Test!
I think they are both great products, and we have many customers using both successfully. You will probably be happy with either. Both have great sound, both work well with Asterisk. The Grandstream is easier to configure, the Sipura has more options. More Grandsreams show up DOA, more Sipuras die in the field. Grandstreams have a few more bugs, but they have much better support. Slight edge to Grandstream on price for similar features. Slight edge to Sipura on build quality. Grandstream is a small and easy to deal with organization. Sipura is Cisco. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 8:30 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Home Test! Hi Michael, so which is your opinion about Sipura and what do you think about Grandstream? I'm looking for opinions of whom has tested the devices and has more experience, not to waste my money. Do you deliver them to Argentina? Erick: spanish ya se que solamente se puede postear en ingles, por eso segui con el dialogo en ingles spanich-off I'm new into this so I appreciate all the recomendations you are giving me. I'm between buying a Sipura 2002 (I didn't know Sipura 200 was replaced) nad a GrandStream HT 486 (or any other model). I have already obtained an FXO port by buying an X100P Clone (here they cost USD10 aprox.), so I want only FXS ports. thanks. 2006/1/23, The VoIP Connection [EMAIL PROTECTED]: We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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 -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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] Home Test!
Sipura and Grandstream are definitely the most popular, but there are others. There is a new IAX adapter with built-in NAT router coming soon that might work for you. Should be announced this week. Contact me if you think you might be interested. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 8:58 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Home Test! I haven't said it but if someone believes there's a better choice than buying a sipura or a grandstream ht, please tell me, I considered thaat two because, here, they are popular. 2006/1/23, Facundo Ameal [EMAIL PROTECTED]: Hi Michael, so which is your opinion about Sipura and what do you think about Grandstream? I'm looking for opinions of whom has tested the devices and has more experience, not to waste my money. Do you deliver them to Argentina? Erick: spanish ya se que solamente se puede postear en ingles, por eso segui con el dialogo en ingles spanich-off I'm new into this so I appreciate all the recomendations you are giving me. I'm between buying a Sipura 2002 (I didn't know Sipura 200 was replaced) nad a GrandStream HT 486 (or any other model). I have already obtained an FXO port by buying an X100P Clone (here they cost USD10 aprox.), so I want only FXS ports. thanks. 2006/1/23, The VoIP Connection [EMAIL PROTECTED]: We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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 -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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] Video Conferencing.
Facundo, If everything goes right, we will be demonstrating an Asterisk based Videoconferencing system at the Internet Telephony expo this week. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 8:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Video Conferencing. I'm looking for point to point Video Conferencing , just because, like I said in other post, I'm doing some tests at homeand I want to try *almost* every feature asterisk has. THank you, I 'll read about it. I also would like to develop for asterisk (it's not for the bounty) but I just don't know much about C or ANSI C. 2006/1/23, Dean Collins [EMAIL PROTECTED]: It's possible to do point to point but not point to multipoint. I tried to get development for this some time ago and no one responded, check out my Video Conference Bounty on www.voip-info.org, since then we have developed our own solution that we have decided to market, it will cost $2,000 for up to 10 users that uses the Macromedia communications server. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Facundo Ameal Sent: Monday, 23 January 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video Conferencing. I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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 -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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] Home Test!
That would be my opinion. However, the additional options on the Sipura may not be of any interest to you. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 24, 2006 8:21 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Home Test! So: Grandstream is easy and Sipura is more flexible and complete. Am I right? 2006/1/24, The VoIP Connection [EMAIL PROTECTED]: I think they are both great products, and we have many customers using both successfully. You will probably be happy with either. Both have great sound, both work well with Asterisk. The Grandstream is easier to configure, the Sipura has more options. More Grandsreams show up DOA, more Sipuras die in the field. Grandstreams have a few more bugs, but they have much better support. Slight edge to Grandstream on price for similar features. Slight edge to Sipura on build quality. Grandstream is a small and easy to deal with organization. Sipura is Cisco. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 8:30 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Home Test! Hi Michael, so which is your opinion about Sipura and what do you think about Grandstream? I'm looking for opinions of whom has tested the devices and has more experience, not to waste my money. Do you deliver them to Argentina? Erick: spanish ya se que solamente se puede postear en ingles, por eso segui con el dialogo en ingles spanich-off I'm new into this so I appreciate all the recomendations you are giving me. I'm between buying a Sipura 2002 (I didn't know Sipura 200 was replaced) nad a GrandStream HT 486 (or any other model). I have already obtained an FXO port by buying an X100P Clone (here they cost USD10 aprox.), so I want only FXS ports. thanks. 2006/1/23, The VoIP Connection [EMAIL PROTECTED]: We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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 -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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 -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --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] Home Test!
We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --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] Need a good extensions.conf sm bus configw/polycom phones
Or on the wiki http://www.voip-info.org/wiki-Polycom+Phones -Original Message- From: Nilesh Londhe [mailto:[EMAIL PROTECTED] Sent: Sunday, January 22, 2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Need a good extensions.conf sm bus configw/polycom phones I am sure that are many more that would be interested (including me). Why not just post it on the list after sanitizing private information? Thanks. On 1/22/06, Max Clark [EMAIL PROTECTED] wrote: I'd love to see this as well. TIA, Max On 1/21/06, Thomas Johnson [EMAIL PROTECTED] wrote: Thanks! I'd love to see your extensions.conf file. I appreciate it. Tom On Jan 20, 2006, at 8:31 PM, Alexander Lopez wrote: Contact me off list, I have a sample extensions.conf file that I can share. It has Paging (one to one and One to Many) Ivr includes, time of da routing and it is geared towards Polycoms. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Thomas Johnson Sent: Friday, January 20, 2006 8:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Need a good extensions.conf sm bus config w/polycom phones Hello- We've got a patched-together extensions.conf that's barely working for us, and we need to get real about using Asterisk. We've got a couple of remote workers with Polycom IP-601 phones, and a single asterisk server, using a couple of incoming DIDs from teliax and sixtel. Does anyone have a good extensions.conf that they'd be willing to share, that provides a real-world tested dialplan? We'd love to see what other people are doing - (preferably those using all these cool features that polycom phones are capable of). Thanks- Tom ___ --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 ___ --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 -- Max Clark http://www.clarksys.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] OT:Snom 360 prompt for registration pwd?
Christian, Why is this this setting on by default? I don't understand why anyone would want this behavior. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Christian Stredicke [mailto:[EMAIL PROTECTED] Sent: Friday, January 20, 2006 8:05 PM To: Colin Anderson Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT:Snom 360 prompt for registration pwd? Did you try to turn Challenge Response on Phone off in the advanced settings on the web interface? CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Friday, January 20, 2006 8:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] OT:Snom 360 prompt for registration pwd? I have a whack of Snom 360's. Occasionally, *some* of them, prompt the user, on the screen, for the registration password. You enter it, everything's OK. Next day, same thing. This is like on 5 or 6 phones out of a lot of 120. It's *always* the same phones. I haven't drilled down to things like firmware rev yet, since I ordered them all as one lot, but I'm wondering if anyone knows under which circumstances a 360 would forget it's reg password? tia ___ --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] Polycom FW
Bill, If you purchased your phones from a certified reseller they should be able to get you these files. If you didn't, contact me off-list. I can help you. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Bill Michaelson [mailto:[EMAIL PROTECTED] Sent: Thursday, January 19, 2006 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom FW Anyone know how to obtain firmware and starter .cfg files for Polycom phones? Despite registering at the Polycom web site, I can't locate this stuff. ___ --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] Recommendations on a WiFi phone for *?
We sell this phone and I like it a lot, but I think Paul is right about wireless in a typical office environment. If, however, you want a phone that you can use in wireless hotspots OR if your office has great 802.11 infrastructure OR if your boss likes to show off his gadgets, then the Zyxel P2000W is a great choice. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Paul Mahler [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 10, 2006 3:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *? I would be MUCH more tempted to use an IAXy or SIP adaptor and a cordless phone. It will be less expensive and it will likely work better. Paul -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joash Herbrink Sent: Monday, January 09, 2006 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *? The zyxel p2000W Works fine, good batt. Live. Decent sound quality. All in all a good product for about 150 euro's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Edelbrock Sent: Tuesday, January 10, 2006 2:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Recommendations on a WiFi phone for *? We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I'm personally drooling over the UTStarcom F3000, but compatibility and shipping ETA info is a bit sketchy. Phil ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006 ___ --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] Grandstream and Snome remote sip stops taking calls
Jason, Your NAT is closing on you, so you need to do something to keep it open. With the snom you can register more often (every minute or so usually works) or you can use QUALIFY. Grandstreams have a NAT keep-alive on the phone which is enabled using NAT TRAVERSAL = YES. This mechanism sends an empty packet at a regular interval to your server keeping the NAT port open. The default keep alive interval is usually fine. Note that if you have more than one phone at a location with you should set USE RANDOM PORT = YES. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Jason [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 04, 2006 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: 'Manny A. Wise' Subject: [Asterisk-Users] Grandstream and Snome remote sip stops taking calls I have remote users that are setup to sip into the Asterisk server. Problem is that if you call there extension after they have been registered For a while there phones don't ring. If I do a sip show peers they can be seen as registered in. Also the user can dial out. If they reset the phone they can receive calls. This seems to be more of an issue with the Grand stream phones. The Grandstream has these two settings I am un sure of. NAT Traversal (STUN): currently set to no SUBSCRIBE for MWI: currently set to no Any ideas? -Jason ___ --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!
And buy your phones from a reputable dealer who will provide you with support. Grandstream's policy (and sipura, snom, polycom, etc.) is to provide warrantee service through their resellers. We have never had them reject a properly documented RMA. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Chris Albertson [mailto:[EMAIL PROTECTED] Sent: Monday, December 26, 2005 11:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stay away from Grandstream! Maybe a better way to say it is Know the limitations of the GS phones and don't try and use them outside of those limits. Don't buy ANY phone you've not tested and used yourself for use by a client. My GS phone has worked fine for years. Even if it were to fail and had to be replaced buying two is still cheaper then one of some of the others. The trick is to use them (or anyhting else) only when you know it will work. That said, the GS 100 is not the best thing to put on a receptionist's desk. I've actually had pretty good luck, even getting to exchangeemail one of thier engineers. --- Elene Kinsky [EMAIL PROTECTED] wrote: We have 2 GXP-2000 dead during automatic firmware upgrade. Devices now send out only one ARP packet for default gateway resolution during boot and nothing more! We've contact Grandstream support, but they cannot help. Now we want to send devices to Grandstream for repair but they on longer reply mail! GXP-2000 was very buggy on attended call transfer, and the problem resolved only after upgrading using latest firmware. Overall GXP is OK, but customer support is terrible. Stay away from them! ___ --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 Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --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] VONAGE and Asterisk
Hi Bret, There is a third option. You can purchase a Business Plus plan. Some details on the Wiki: http://www.voip-info.org/wiki/view/Asterisk+and+Vonage Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: trixter aka Bret McDanel [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 06, 2005 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VONAGE and Asterisk On Tue, 2005-12-06 at 20:08 -0400, Dakota wrote: Can Vonage work with Asterisk? Yes under 2 different plans. 1 is to get the SIP softphone plan which is a different phone number and limited in the number of minutes per month you can use. 2 break into your ATA and get your credentials so you can register directly. This way however has a major drawback in that it is a violation of their TOS. Due to case law in America (where vonage is so it doesnt really matter where you are) a violation of the TOS is a felony pursuant to 18 USC 1030(a)(5) punishable upto 10 years in jail and fines upto $250,000. While it is unlikely to result in anything other than them canceling your account if they discover it, you should be warned of the downside on that should they choose to persue it. I dont advise you to try this for that reason. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --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] Re: ip phone
Listen, we sell a lot of Budgetones so I'll admit to having an agenda here, but I like to think I'm pretty honest and objective about product advice. These phones got a bad rep based on early versions which admittedly had many problems. They now work quite well when configured properly. No, they are not as good as a $200 Polycom (gasp!). As far as being the equivalent of a $10 k-mart phone, I'd like to see a $10 phone that has any of these features - let alone ALL of them: backlit display, blind and attended transfer, three-way calling, off-hook auto-dial, auto-answer, speakerphone AND headset jack, custom ringtones (including one that announces the caller ID), time and date display, the list goes on. This phone is not suitable for all applications but it is not fair to say that it doesn't work well or offer great value for the money. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Friday, November 18, 2005 8:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: ip phone Maybe grandstream budgetone 100 series will fulfill your requirement. It's very good for such a cheap sub-50 phone. We have two of these and they are the VoIP equivalent of a $10 K-Mart phone. I won't even use them in my house, much less the office. Might be carefull with assumptions in this area... depending upon where you are from and what type of service one is accustomed to using (or receiving), the term quality has as many interpretations as there are countries (or counties in some cases) in this world. Some would consider the 100 series as a significant improvement over what they currently have for service, while many others would consider it close to the bottom of the stack of sip phones. I'm not trying to defend anyone's opinion or propose alternatives. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Snom clients deregistering
There is a setting on the Advanced page called Challenge Response on Phone. Turn this setting to Off and your problem will be solved. Also, we usually set the Proposed Expiry to 1 minute On the SIP page when phones are behind a NAT. -Mike -Original Message- From: Richard Watson [mailto:[EMAIL PROTECTED] Sent: Monday, November 14, 2005 8:30 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Snom clients deregistering Michael Crown wrote: Does the phone ocasionally prompt the user for a password? -Mike Yes it does How did you know? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Cisco 7960 Skinny Firware
We can get it for you.Contact me offlist. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Bobby Lacey [mailto:[EMAIL PROTECTED] Sent: Sunday, October 30, 2005 3:15 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Cisco 7960 Skinny Firware Hello, I have just acquired my first 7960 from a business sale. It has already been preloaded with the 7.3 SIP image which works flawlessly with my Asterisk box. I want to experiment with chan_sccp and therefore I would need the skinny firmware = 7, I guess. Could someone tell me an outlet where I could purchase a Smartnet contract to download this firmware? I have been unable to find a retailer online that can help. Thanks in advance for any help. BL ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] The VoIP Connection has $$$ opportunities for Asterisk experts
The VoIP Connection is growing and we have opportunities for talented individuals with Asterisk and general VoIP experience. We have a need for assistance with the following activities: 1) Level one and two customer support 2) Asterisk custom application development 3) Asterisk product development 4) VoIP device configuration and troubleshooting (Grandstream, Sipura, Snom, Polycom, Cisco, etc.) 5) Asterisk system design and deployment We have openings for full-time positions at our Florida facility as well as opportunities for off-site consultants and support engineers on an hourly, part-time or per-project basis. Compensation will vary according to experience, activity, and type of engagement. A baccalaureate degree is not a strict requirement. Preference will be given to pragmatic, results oriented individuals possessing a track record of success with real world applications. Candidates that do not demonstrate a high level of professionalism and strong communications skills will not be considered. Experience configuring and supporting the VoIP Connection VS-1 Asterisk Server is a major plus. Interested professionals please send resume and references to [EMAIL PROTECTED] If you are attending Astricon we would love to meet you in person. Visit us in Booth #106 or in the Code Zone. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Asterisk overheating on VIA Epia M Series motherboard
The EPIA M runs fanless at 600Mhz. If you are running it at 1Ghz you need a fan. http://www.viavpsd.com/product/epia_m_spec.jsp?motherboardId=81 Having said that, Asterisk can overwhelm these boards unless you run a very lean distro and configuration. They are beautiful little machines but they are designed for maximum reliablility at a relatively modest level of performance. Codec transcoding is not possible for more than a few channels. Running a full PRI on a Digium TE110 is pushing it, especially with echo cancellation. It should handle a TDM04B just fine though. We run Fedora Core 3 on these with no problems. You could try running top to see which process is chewing up your cycles. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Angus Comber [mailto:[EMAIL PROTECTED] Sent: Monday, September 05, 2005 4:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk overheating on VIA Epia M Series motherboard Hello I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink. Currently system is running off standard IDE hard drive - because I couldn't get astlinux to run with my Digium TDM04B card (only PCI card in system). Strangely I also have the same system also running SUSE Linux running as a file server and that does not run so hot and does not overheat? Why the difference? Just booting up both systems for 15 minutes you can tell the Asterisk box is quite a bit hotter. Also the Asterisk box overheated (well think that was the problem) and stopped operating as PBX at one stage. Anyone any experience of this sort of thing? any ideas how to fix - ideally I don't want to have to fit a fan. Is SUSE not the best distro to use for this sort of thing? Should it be something to take up with VIA? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Dell 2850 anyone ...
Hi Bill, We just built one for a customer with Fedora Core 3 and a TE210. We get PCI parity errors and the machine shuts down. I'm sure we'll get it working, but it hasn't exactly been the smoothest install ever. I agree that the second CPU and GB of RAM is probably overkill and as you know, I also share your bias towards two smaller servers as opposed to one big one. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: William Boehlke [mailto:[EMAIL PROTECTED] Sent: Thursday, August 25, 2005 5:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell 2850 anyone ... We successfully use 2850s with Digium T1 cards, though I don't think we've installed a TE411P. It'll handle two T1s with ease. You don't need the second processor or the second GB of RAM for the expected load. For your configuration we would usually use two single processor 1u servers with RAID 1 for roughly the same cost so we're not vulnerable to a motherboard failure. William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Bunch Sent: Thursday, August 25, 2005 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dell 2850 anyone ... Can anyone comment or share experences with using Dell 2850's with Asterisk. Proposed config is 2850, 2 x 3.6g procs, 2 g's of ram, 4 x 36g 15k rpm drives raid 10, Digium TE411P ( the echo cancelling cards ). Expected load is 1 or 2 pri's (most likely 1 ) 100 Polycom phones on the local network, 15 phone on a remote T1. 6 phone remote via the internet using IAX, Voicemail for 125 users. As little transcoding as possible. G.729 licenses. If Dell is not the answer how about sharing what works. I do need a natinal brand that I can take to managment. Alan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.15/81 - Release Date: 8/24/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.15/81 - Release Date: 8/24/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Load Testing
Anton, A great tool for ghetto call capacity testing is a single snom phone. There is no limit to how many calls a snom phone can make, just put it on hold and dial again. So, with a single snom phone and a little imagination you can test any number of scenarios. You can approximate basic SIP capacity by creating an extension that plays the asterisk test message and dialing it repeatedly until quality starts to degrade or asterisk gives up. To simulate actual call throughput you really need another (faster) machine to connect to, but you can use the same technique. You can run top on the console while you are doing your tests to see what resources you are using. Check your logs when you are done to see what errors were generated when it came unglued. CPU is not always the limiting resource, especially with Digium card interfaces which tend to be bound by FSB speed, but echo cancellation and codec conversion will burn a LOT of cycles. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Friday, August 12, 2005 9:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Load Testing Guys. How and which tools to use to load test an asterisk install? Say for example, you need to see how many calls can be routed thru before losing quality and making the cpu jump to the roof? ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Supervised transfer problem with BudgetTone
Section 4.3.7.2 from the Bugetone Manual: The user can transfer an active call to a third party with announcement. The user presses the flash button and hears a dial tone, then dial the 3rd partys phone number followed by pressing send button. If the call is answered, press flash to complete the transfer operation, if the call is not answered, pressing flash button to resume the original call. Notes: If attended Transfer fails, the BudgeTone phone will ring the user to remind that another party is still on the call, the user can then pick up the call using handset or speaker. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] Sent: Thursday, August 11, 2005 5:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Supervised transfer problem with BudgetTone [EMAIL PROTECTED] a écrit : On Thu, 11 Aug 2005, Nicolas Schmerber wrote: All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Hi, You'll need to switch to the CVS-HEAD version of Asterisk in order to have supervised transfers. Steve ___ Asterisk-Users mailing list Asterisk-Users@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 When looking at a recent firmware changelog of Grandstream , it says BT should support supervised transfer, so shouldnt it work ? ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Supervised transfer problem with BudgetTone
Nicolas, Just did some quick testing and the instructions are incorrect. You need to press transfer to complete the transfer instead of the second flash. This actually makes more sense. Attended and regular transfer both work perfectly with the following settings: Enable Call Features: Yes Disable call Waiting: No Send Flash event: No DTMF should be whatever * is set to, but in-band won't work properly if your codec is anything other than U-Law. By the way, the newest firmware also makes the long overdue conference feature work properly. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] Sent: Thursday, August 11, 2005 10:41 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Supervised transfer problem with BudgetTone The VoIP Connection a écrit : Section 4.3.7.2 from the Bugetone Manual: The user can transfer an active call to a third party with announcement. The user presses the flash button and hears a dial tone, then dial the 3rd partys phone number followed by pressing send button. If the call is answered, press flash to complete the transfer operation, if the call is not answered, pressing flash button to resume the original call. Notes: If attended Transfer fails, the BudgeTone phone will ring the user to remind that another party is still on the call, the user can then pick up the call using handset or speaker. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] Sent: Thursday, August 11, 2005 5:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Supervised transfer problem with BudgetTone [EMAIL PROTECTED] a écrit : On Thu, 11 Aug 2005, Nicolas Schmerber wrote: All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Hi, You'll need to switch to the CVS-HEAD version of Asterisk in order to have supervised transfers. Steve ___ Asterisk-Users mailing list Asterisk-Users@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 When looking at a recent firmware changelog of Grandstream , it says BT should support supervised transfer, so shouldnt it work ? ___ Asterisk-Users mailing list Asterisk-Users@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 Tried this manipulation a few minutes ago : A calls B , B pushes flash button ( A is waiting with a mp3 played) B calls C pressing Send ; C answers B presses flash button again ; C is so on hold (with a mp3 played) B hangs up But A and C arent in connect ; the phoneof B rings ( to tell someone is in wait : A) So it seems to fail What should i put in grandstream config for the next item : /Enable Call Features: Y/ N ? //Disable Call-Waiting: Y/N ? //Send DTMF: / in-audio / via RTP (RFC2833) / via SIP INFO /Send Flash Event: Y / N ? / Any others Ideas ?. Thx Nicolas S. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] gxp-2000 tftp cfg
hi,can you someone post tftp template for gxp-2000?like http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txtthanks---Marek Cervenka=== It's here: http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/gxp2000_config.txt Michael Crown Managing Partner www.thevoipconnection.com sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition
This is a very interesting converation, but it seems like the BIZ forum might be more appropriate... Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Lee Howard [mailto:[EMAIL PROTECTED] Sent: Monday, June 13, 2005 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition Andrew Kohlsmith wrote: On Saturday 11 June 2005 19:51, Lee Howard wrote: I don't think that lack of mindshare completely defines the reasons behind Asterisk fork failures. It places all of the blame on the forkers. I think the truth, though, is that they not only fail due to lack of mindshare but also due to competition from Digium's own Asterisk community. Forks are not succeeding, yes, but Digium has a hand in that... of course they do. I'm not saying you're wrong, but I'm curious: how does Digium have a hand in a fork failing? That's what I tried to explain in my last post, in particular after this first statement. Forks enter a hostile competition rather than a healthy competition. I've heard more talk about Asterisk forks than I've ever heard about forks of any other other open-source project. I think that this says something about how difficult-to-swallow Digium's dual-license decree is for a lot of prospective contributors/developers. I disagree; if it were that hard to swallow the project would either be 90% digium-written (it's not) or it would be a total flop (again it's not). If you (or someone else reading this post) is in a position to give statistics on what percentage of the code is Digium-written (or Digium-rewritten - in the case where a disclaimer is not obtained for some unpatented work and Digium rewrites the work independently) then I would be thrilled to see it. We see this happen all of the time with the Linux kernel. It happens with HylaFAX. It happened with X. I'm sure it happens a lot with many other open-source software projects. It happens easily and usually is a healthy process because the playing field is even. Agreed. But where are the successful Asterisk forks? I don't know of any successful Asterisk forks (unless http://www.asteriskwin32.com is considered successful - although I'll admit that I'm not really in-the-know). But this was my point: that the way things were set up by Digium makes a successful fork difficult. Digium always has an upper-hand, and things were set up intentionally this way. Again, I don't take particular issue with this. I'm just trying to explain why forking Asterisk would not be a particularly easy task. Of course, this healthy forking cannot be done with Asterisk because Digium will not accept any non-disclaimed code into their repository. ... What you'd described about distribution-maintained patches has nothing to do with this. Digium could take a distribution-maintained patch and rewrite it into Asterisk proper under the dual license (as could any other contributor) and you'd still gain the benefit of the patch. I'm not sure I see where you're going here. If you (or someone else reading this) has the necessary information to provide statistics on how what percentage of the code comes from rewrites of non-disclaimed code, then I would be particularly interested in hearing it. I suspect, though, that it is a rather small - perhaps insignificant - amount. But, yes, providing that there is not a patent involved - yes, the work could be rewritten and integrated. But this was my point: that given the right environment forks can benefit from each other. The one thing that an Asterisk fork can never do, though, is relicense itself. Only Diguim can do that. If Digium had wanted an equal footing in this regard then Asterisk would be LGPL or BSD or something a bit more liberal. So if I'm a manufacturer of PBXes and have some proprietary IP that I do not wish to be GPLed, then if I want to use Asterisk somehow, then I can really only work with Digium for licensing. All of the other forks will be license-prohibitive. I have to admit that I know quite a few people with their own modules and such to replace what they feel is bad code and just won't contribute it back to Asterisk due to the friction they've received about the patch. I, on the other hand, tend to bitch loud and continuously enough and wear them down to the point of accepting it. :-) So we're not in disagreement, it would seem. Getting code contributions into Digium's Asterisk codebase is not something that many average people are going to want to undergo. From what I've seen, friction is a bit light of a term for it. It seems much more
RE: [Asterisk-Users] GXP2000 and hint LED's
That is the entire package as it was submitted to us from Grandstream. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: Friday, June 10, 2005 1:46 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] GXP2000 and hint LED's On Thu, 9 Jun 2005, The VoIP Connection wrote: This is supposed to be the final version: http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/GXP2000_Rel ease_1 .0.1.9.zip Have you received an updated tftp config template as well? We asked for and received one with a 1.0.1.9 early beta version. Peter ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Asterisk Live! CF
We use Via EPIA Mini-ITX boards extensively and have had NO problems, with interrupts or otherwise. I can't vouch for or defend any of the Pentium mainboards you reference, but the single board machines we use run Asterisk very reliably. We don't load them up with a bunch of peripherals or exotic graphics cards, but I don't know why anyone ever would. It is true that they will not run gcc i686 binaries reliably. Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Kanuri, Seshu (Company IT) [mailto:[EMAIL PROTECTED] Sent: Thursday, June 09, 2005 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Live! CF Abel, I am working on Intel boards only. I have tried VIA boards and I do not recommend anyone to work on VIA boards for a production system. The reasons for this being that there are just way too many issues with these boards, gcc being just one of them. The main issue is Interrupt Conflicts and incompatibility for many accessories. The link below has more information on these problems: http://pcbuyersguide.com/hardware/motherboards/VIA-Problems.html A few more snippets are here http://www.georgebreese.com/net/software/ Kris probably will answer your other question. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of abel Sent: Thursday, June 09, 2005 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Live! CF Seshu, Are you working on a VIA based motherboard? I am working on a VIA based motherboard. Andy Powell (author of Asterisk Live! distro) tells me that VIA is not quite good when emulating i686 behavoir and since his distro is compiled for i686... We are trying to confirm that but may be interesting to know about your setup and how is Kristian's distro compiled. On Mon, 6 Jun 2005 16:41:43 -0400, Kanuri, Seshu (Company IT) wrote Kristian, I am talking about your distro, that does not seem to be able to boot when I have mounted (if that is the right word) the CF into my Dell Server and tried to boot from it as the only IDE drive available. The Linux just does not kick in. If you want to debug this I can Fedex to you, my 800MB CF disk with your distro on it, you for your RD. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Monday, June 06, 2005 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Live! CF abel wrote: My theory is that the 64 MB image is built with a specific hdd form factor and when burning onto a different size CF it is mapped differently and it does not work. On the other hand, you always can find out how the device is beeing seen by the system and customize the binary image accordingly. Other software prepared to be run from CF (I recall WISP, the LEAF branch for wireless routers) have a final step which takes the software already compiled and 'packages' it to build the disk image. I would be extremely happy if I could download the code tree and run that final step by myself to get the disk image that suits my needs. Second best would be to get the source tree and compile all the stuff to get that point. Is that possible? Is the code available in the way I need for this operation? TIA. abel, This is simply untrue. My distro's (AstLinux) 32mb CF images work on anything... http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid= 3 -- Kristian Kielhofner NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
RE: [Asterisk-Users] GXP2000 and hint LED's
This is supposed to be the final version: http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/GXP2000_Release_1 .0.1.9.zip Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: Thursday, June 09, 2005 3:08 PM To: Julian J. M.; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GXP2000 and hint LED's On Thu, 9 Jun 2005, Julian J. M. wrote: I've just checked the download page, and the latest firmware available is 1.0.1.8. Where did you find 1.0.1.9? This phone has some nasty bugs, one of them being that the other end HEARS you after you press the Transfer button and you hear a dialtone. It doesn't send any message to asterisk so that it can play music on hold to the caller. It is a pre-release version, not the actual 1.0.1.9. We received it to test a fix for a problem we observed in 1.0.1.8. So far we have not encountered any bugs with this pre-release. You may want to ask Grandstream support when it will be released. Within 24 hours of us reporting new bug we received a firmware which fixed the problem. Peter ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] GXP2000 and hint LED's
James, If you don't think you want to wait for the Grandstream, the snom 360 will do what you need (12 programmable buttons). We are offering great pricing on these right now. http://www.thevoipconnection.com/store/catalog/product_16234_snom_360_Execut ive_IP_Telephone.html Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: James Bean [mailto:[EMAIL PROTECTED] Sent: Thursday, June 09, 2005 5:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] GXP2000 and hint LED's Did that pre-release version fix that bug where the other party can hear you when you pressed the transfer button ? Does it also enable the leds next to the speeddial buttons like the snoms ? Unfortunately not, Grandstream didn't admit to me that they were going to program the LED's like the snom SUBCRIBE/NOTIFY, they told me the LED's were additional incoming line indicators, not LED's for the function keys to be programmed. Which is a little stupid, if they don't do the LED's like the snom then the phone is really no better then the BT102, just with a bigger LED and multiple sip account capability. If you want the 1.0.1.9 firmware pre-release goto www.atp.org.au and on the main page near the bottom it gives you a link. Peter seems to be on the ball more then me about these phones as grandstream gave me the standard replies, Peter do you know for sure if grandstream have a timetable for the function led's cause I need to rollout about 50 phones and need 6-7 led's for display, which means a snom220+expansion, and gxp2000 seems perfect if it worked. James ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Books
We have it: http://www.thevoipconnection.com/store/catalog/product_16198_VoIP_Telephony_ with_Asterisk_by_Paul_Mahler.html Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: John H [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 08, 2005 12:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Books Hello all, I was wondering if anyone know where i can find a book on Asterisk, i have been told about VoIP With Asterisk but i am unsure where to find it, any ideas plase? ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] secretary function
Christian, Follow this procedure to make sure your firmware is up to date: In the advanced setting menu set Update Policy to Ask for Update In the Software Update menu set firmware to http://www.snom.com/download/share/snom360-3.60i-SIP-j.bin Press the Load button. When the phone prompts for Update New Firmware press Check. DO NOT UNPLUG THE PHONE BEFORE PROCESSS COMPLETES OR YOU WILL HAVE AN EXPENSIVE GERMAN PAPERWEIGHT. For a blind Transfer: Press Transfer, enter extension or press speed dial to transfer to, press Check. If more than one call is active, use arrow keys to select the caller you want to transfer, press Check again. For an attended transfer, follow the steps Julian outlined. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Julian J. M. [mailto:[EMAIL PROTECTED] Sent: Friday, June 03, 2005 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] secretary function Try this: 1) You're on a call 2) Push a Line button, so that you get dialtone 3) Dial the boss extension # 4) Hey boss, you have a call from XXX 5) Push Transfer 6) You can select which call to transfer (if you have more that 1 on hold) 7) Push transfer again. Julian. On 6/3/05, Christian Hiller [EMAIL PROTECTED] wrote: Hello, we got a SNOM 360 here and this gota TRANSFER button. With this i can transfer a call from my phone another one. But when i push this Button and transfer the call to another phone, i get kicked out. Now, every secretary first asks the chief if he is available or not - how can i implement this feature ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] VoiPSupply Dot Com
I know I'm running the risk of fanning the flames on an already belabored thread here, but there is some misinformation flying around. Credit card fraud is an unfortunate fact of life, and it costs everyone who isn't perpetrating it money. There is no single universally agreed on process that will guarantee a merchant protection. If there was, somebody would figure out how to game it. Different banks have different merchant account requirements, e-businesses use different procedures to protect themselves, and of course different businesses tolerate different levels of fraud. Some vendors require that items be shipped to an address on file to protect themselves. Others (like us) do not. We have a process for validating the card for these cases which our bank has agreed is adequate in most cases. It's a little more time consuming but it is something that many of our customers require. There was a misunderstanding, let's move on. I am really tired of seeing VoiPSupply Dot Com every time I open a digest email... Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Thursday, June 02, 2005 8:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoiPSupply Dot Com He is right Karl. Without the ship-to being on file with the bank.. the company can be held responsible for fraudulant purchases. On 5/31/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: I'm amazed that this thread keeps going... About the claim of Ship-To being on file with bank... CDW doesn't have a problem with it... Ingram Micro doesn't have a problem with it. Merisel doesn't have a problem with it. Digi-Key doesn't have a problem with it... Why would Voip-Supply??? We accept packages every day with the same Ship-To address specified to Voip-Supply... Additional comments dispursed throughout At 02:32 PM 5/27/2005, you wrote: On 5/27/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: At 08:59 AM 5/27/2005, you wrote: [ snip for brevity ] I just wanted to clarify ... this isn't a voipsupply.com problem at all, butrather a courier screwup... which happens anywhere and at anytime... right? TWO screw ups in the shipment. 1.) It was shipped to the Bill-To address. Since there is no one there during the day I had to sit and wait for it lest it not be delivered. This screw up has to do with the person that ordered it, because they didn't have the ship to address on file with their bank. This was not a paypal transaction. The PO had BIG BOLD LETTERS - Ship To: I'm unaware of any practices with the bank that requiring Ship-To addresses to be on file with them. Perhaps your financial institution is a bit different? 2.) when an order is placed on a Tuesday AM (or) Monday PM, and it's priority overnight, and it's across town, and the tracking number was supplied on Wednesday one would expect that it would show up Thursday, not Friday. See above, again this is a screw up that happened because of the one that ordered it, by NOT having the ship to address on file with their bank. Where do you get this Ship-To on file w/ Bank idea? Anyhow, you were already answered before that it had to do because YOU didn't have the address on file with your bank. Why are you repeating this lie that it is voipsupply.coms fault? Be repeating it you make yourself look more like a politician or media person, but certainly not someone that is in the electronic engineering business. No I will not believe it because I read it twice, so stop it. No lie... Fact. There is a difference... So, what we have here is one problem compounded by another, none on behalf of the courier. Exactly, but on behalf of the ordered. If you give a Ship-To address that is NOT on file with your bank, you will NOT get it to that address, and it WILL delay shipping. Gosh dang spin doctors... Where does it state this??? Prove it. Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Bethesda MD: (301) 576-3014 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or
RE: [Asterisk-Users] VoiPSupply Dot Com
I understand all that. I just wanted to try and clear up some of the confusion with respect to credit cards and merchant accounts in hopes that it might save somebody from some frustration in the future. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Neal Walton [mailto:[EMAIL PROTECTED] Sent: Thursday, June 02, 2005 12:53 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] VoiPSupply Dot Com Karl has already stated more that once that this was NOT a credit card purchase. If a credit card was not used for the purchase, why would you need a Ship To address on file with the credit card company? Cory Andrews from VOIPSupply has also admitted that the sales rep who took the order made a mistake and failed to notice that a Ship To address had been supplied. On Thursday, June 02, 2005 9:04 AM, The VoIP Connection [SMTP:[EMAIL PROTECTED] wrote: I know I'm running the risk of fanning the flames on an already belabored thread here, but there is some misinformation flying around. Credit card fraud is an unfortunate fact of life, and it costs everyone who isn't perpetrating it money. There is no single universally agreed on process that will guarantee a merchant protection. If there was, somebody would figure out how to game it. Different banks have different merchant account requirements, e-businesses use different procedures to protect themselves, and of course different businesses tolerate different levels of fraud. Some vendors require that items be shipped to an address on file to protect themselves. Others (like us) do not. We have a process for validating the card for these cases which our bank has agreed is adequate in most cases. It's a little more time consuming but it is something that many of our customers require. There was a misunderstanding, let's move on. I am really tired of seeing VoiPSupply Dot Com every time I open a digest email... Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Thursday, June 02, 2005 8:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoiPSupply Dot Com He is right Karl. Without the ship-to being on file with the bank.. the company can be held responsible for fraudulant purchases. On 5/31/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: I'm amazed that this thread keeps going... About the claim of Ship-To being on file with bank... CDW doesn't have a problem with it... Ingram Micro doesn't have a problem with it. Merisel doesn't have a problem with it. Digi-Key doesn't have a problem with it... Why would Voip-Supply??? We accept packages every day with the same Ship-To address specified to Voip-Supply... Additional comments dispursed throughout At 02:32 PM 5/27/2005, you wrote: On 5/27/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: At 08:59 AM 5/27/2005, you wrote: [ snip for brevity ] I just wanted to clarify ... this isn't a voipsupply.com problem at all, butrather a courier screwup... which happens anywhere and at anytime... right? TWO screw ups in the shipment. 1.) It was shipped to the Bill-To address. Since there is no one there during the day I had to sit and wait for it lest it not be delivered. This screw up has to do with the person that ordered it, because they didn't have the ship to address on file with their bank. This was not a paypal transaction. The PO had BIG BOLD LETTERS - Ship To: I'm unaware of any practices with the bank that requiring Ship-To addresses to be on file with them. Perhaps your financial institution is a bit different? 2.) when an order is placed on a Tuesday AM (or) Monday PM, and it's priority overnight, and it's across town, and the tracking number was supplied on Wednesday one would expect that it would show up Thursday, not Friday. See above, again this is a screw up that happened because of the one that ordered it, by NOT having the ship to address on file with their bank. Where do you get this Ship-To on file w/ Bank idea? Anyhow, you were already answered before that it had to do because YOU didn't have the address on file with your bank. Why are you repeating this lie that it is voipsupply.coms fault? Be repeating it you make yourself look more like a politician or media person, but certainly not someone that is in the electronic engineering business
RE: [Asterisk-Users] handytone 486
The Handytone 488 does this. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 31, 2005 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] handytone 486 Betül Gözlükoglu wrote: Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... On my 486 I can't dial out on the FXO port, it's just a lifeline. There are rumours that there is an update or a new version that can do this. The SIPURA 3000 supports this and work with Asterisk. /Olle Astricon - the Asterisk User's conference - Madrid June 15-17 http://www.astricon.net/europe/ - Register today! ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Grandstream GXP-2000 headset
It's very hard to find a headset that will fit this phone. We managed to find an adapter that will allow you to use almost any mini-plug headset. We will be including this adapter with every GXP-2000 purchased from www.thevoipconnection.com. If you need one, contact me off list and we will send you one. Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: marek cervenka [mailto:[EMAIL PROTECTED] Sent: Monday, May 23, 2005 2:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream GXP-2000 headset Hi all What headset do people use with the GXP-2000? Any recommondations for or against particular models? i'm sent mail to [EMAIL PROTECTED], need info too btw i'm asked that will support IAX, they respond yes, if customers want it - write them --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] * Server
Chris, Obviously we can't publish a list of our customers on this or any other news group, but if you would like some references we would be happy to provide them. I know some of them are on the list, maybe they will be kind enough to share their opinions. The VS-1 has been performing flawlessly in production at numerous locations for over a year now, and it should continue to do so for many more. The reason we don't specify an Asterisk version on our web site and data sheet is that the marketing hype for the VS-1 pre-dates Version 1.0. The latest VS-1 comes with two versions of Asterisk installed: One is a stable version (currently 1.0.7), the other is a development version which is built from CVS head and the occasional assorted patches. The management interface allows the administrator to easily select which of the two versions of Asterisk (stable or development)they wish to run. Either version can be updated or modified by the administrator should they choose to. Since the door has been opened, I'll offer up a little more hype: Unlike some other turn-key products, the VS-1 is not an attempt to dumb down or obfuscate Asterisk. It is not impaired or restricted in any way. Our web management interface is a layer on top of the standard configuration file interface and it does not interfere with or overwrite direct edits to the files. The system comes with most of the popular functionality pre-configured: voicemail, festival, moh, meetme, FOP, etc. and also includes a number of other tools and utilities that ease setup, phone provisioning and remote administration. Like a lot of good technology, the VS-1 is 1% inspiration and 99% perspiration. A significant amount of engineering has gone into this unassuming little black box. It is designed for stability and reliability as opposed to ultra high performance, but with limited transcoding it can easily handle a full T1 of PRI. It's also pre-configured for several popular VoIP service providers and it gets along just fine with a TDM400P card. We stand behind it with a one year replacement warrantee and 30 day money back guarantee. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: snacktime [mailto:[EMAIL PROTECTED] Sent: Thursday, May 12, 2005 4:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * Server On 5/12/05, Montague, Clarence [EMAIL PROTECTED] wrote: Any reviews/comments out there on this server? Looks solid.. But would like to know if anyone has purchased one of these before. Any other companies out there offer pre-built * servers that someone would like to comment on? http://www.thevoipconnection.com/store/catalog/product_16214_VS1trade_ Asterisk_Voice_Server.html Personally I would want to see the full specifications and get some more information about it's track record in production use. If this unit was vouched for by some recognized names that have used it in production, and if they stated what version of asterisk was used instead of just saying it's their own 'certified' version, I might be inclined to say it looks like a good deal for a small business or office environment. As is it seems short on details and long on marketing hype. Chris ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] * Server
Adrian, Strictly speaking, you are correct. Asterisk is not a true proxy server and SER is not currently included. -Mike -Original Message- From: Adrian A [mailto:[EMAIL PROTECTED] Sent: Thursday, May 12, 2005 11:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * Server It's mentioned in the description that: The VoIP ConnectionT Asterisk Voice Server combines the functionality of a PBX, SIP proxy, Voice Mail server, and more. As far as I know, Asterisk does not act as a proxy. Does it have SER included or is it just a confusion of terms? ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] * Server
-Original Message- From: snacktime [mailto:[EMAIL PROTECTED] Sent: Thursday, May 12, 2005 11:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * Server That certainly sounds a lot better than most of these outfits selling the cheap dell SC420's with asterisk thrown on. We think so, but I guess they have their place as well. I would think a higher end server with the same type of warranty and testing would also be in demand, given the number of people who aren't really sure what hardware works well with asterisk. More models are on the way. We also offer a hosted version of the VS-1, the VS-X. Have you used it with any of the E1/T1 cards? Chris The VS-1 works beautifully with a TE110 card. Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] A good SIP receptionist phone
Christian, The current snom scheme is great for most applications and would probably work reasonably well for this user. If you read the original post, he indicates that he would be happy with a snom if he could make it work, and I think this is the main issue with the snom 220 - getting this setup to work can be a little tricky. We have found in the past that extension monitoring and multiple registrations don't play well together, which makes it hard to use for a lot of situations. This may be fixed now, I'm not sure when we last tested this. Receptionists who are used to the usual key system park and page routine can be trained pretty easily to transfer to extensions if the system is set up right. In my experience, most of these people are not stupid. Managing and routing an endless stream of incoming calls is challenging and stressful even under ideal circumstances. When a system doesn't work the way it should it can be very frustrating. I know this logic is kind of inside-out, but if you think of a receptionist as a human auto-attendant/IVR and design a phone that supports this role you will sell a lot of them. A lot of times the receptionist (i.e. office manager) is the decision-maker for phone system purchases. Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Christian Stredicke [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 03, 2005 1:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Olle E. Johansson Subject: RE: [Asterisk-Users] A good SIP receptionist phone We at snom would love to have a good LED integration with Asterisk. The current state seems to be a good start, but can use some improvements. What would be the best way to push this? Maybe sit together for a few days and work on the integration (doing some dirty hacks). Who would be the right person to talk to? Olle? CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, May 02, 2005 10:46 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] A good SIP receptionist phone Adam Goryachev wrote: On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote: 2) There isn't anything like what you want. I know, I want the same thing. There is no phone out there that will do this with any protocol that asterisk uses. This is the one major failing of asterisk ( and voip in general. I smell an oportunity for a phone manufacture ), and what keeps it out of a lot of places. It's alright, you can come out from under your rock now The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) phones can all do what he wants. ie, have multiple lines with blinking red lights when a call arrives on that line. The polycom ip600 and cisco 7960 both have 6 lines available. Regards, Adam Ok, this is the first I've heard about it. Will the lights show call status? As in, if the call is put on hold on one of those other extensions, it will flash? Or go green ( or another color ) when a call is connected on another extension? Basically a mimic of the partner ACS systems? To my knowledge, there is no such thing. Am I wrong? Sean ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Grandstream ATA 286 problems
We have sold a lot of these adapters and we do have a few problems with them, but for every one that has problems there are at least hundred that work perfectly. Do we wish that they all worked perfectly? Of course. Luck of the draw I guess. Grandstream products have a one year warrantee. If you can show (with invoice or otherwise)that your product is in warrantee we will exchange it, regardless of where you bought it. All we will charge you for is shipping. Send an e-mail to [EMAIL PROTECTED] for RMA instructions. Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Andrejus Stavickis [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 26, 2005 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grandstream ATA 286 problems Hi, Well, in my case I have a 486 and a hell lot of the problems with that. First I have not being able to use features like call transfer or anything like that (built-in ones) - it will just not respond to those commands. And Grandstream sent me a reply to my problem saying: this is the problem with your * configuration and those features are not supposed to work without Asterisk. But eventually with the latest firmware I've managed to make it work after some voodoo. But the new firmware introduced another issue, which is kind of weird: my ADSL modem will resync every 10-15 min if I connect Grandstream ATA 486 to PSTN (do not get me wrong, I really put the ADSL filters in). As soon as I remove ATA 486, ADSL stays solid and does not resync. In your case you would just get a bounce backs from grandstream to your vendor and back, but in my case vendor will just not respond at all to any communication means. So beware VOIPSUPPLY.COM seems to be a bunch of funny people who will not stand behind the products they sell. So in my case I not just made a worst purchase, but also choose a worst supplier. Sincerely, --Andy x6722 I contacted the vendor I bought it from, and they said to contact Grandstream. I contacted Grandstream, and they told me to hit refresh in my browser After sending them the Ethereal trace, I haven't heard back from them yet. I think it's the worst purchase I've ever made. On 4/25/05, Anton Krall [EMAIL PROTECTED] wrote: Anobody had any problem with GS ata 286? The past few days Ive been having some problem with it, while making a call or during a call, I suddely hear a low noise like a car engine starting and then the ata dies, as if it got stuck or frozen. Anybody had these problems? ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?
That's a new one. Occasionally they show up dead, but usually if they work the sound quality is excellent. I'll forward this on to Grandstream. In the meantime, please post it to the newsgroup. -Mike http://www.thevoipconnection.com/forums/index.php?board=4.0 Michael Crown Managing Partner The VoIP Connection www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Thursday, April 21, 2005 3:00 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet? Have you had some experience with GS ATA 286? I have 2 analog phones connected and using the latest firmware on the ATAs and from time to time, while in a call, the line just gets filled with line noise and you have to hit flash and then flash again to retake the call and the line noise is gone Do you know what might be causing the problem? I do noticed that 286 ATAs run very hot. Other than that, you sometimes need to reboot the ATA once a day but nothing worth fighting for. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of The VoIP Connection Sent: Miércoles, 20 de Abril de 2005 11:29 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet? We have had these for a while now and they work great with Asterisk. As Brian said, setup is a breeze. We have not experienced any of the audio issues he describes. They ship with an early (somewhat limited) firmware version so an immediate upgrade is probably in order (set tftp server to 168.075.215.189). Obviously, it may take a few versions to get everything perfect, but it's pretty darn stable for a new product. It has all the call control features you'd expect from a business phone plus it has a backlit display AND built-in Power over Ethernet AND 10/100 switch AND 4 independent registrations AND 7 programmable speed-dial buttons. One strange thing - the headset jack is 3.5mm and it's pretty hard to find headsets with that size jack. We did find some adapters to convert to the usual 2.5mm. US$114.95 at https://www.thevoipconnection.com/store/catalog/product_16233_Grandstream_GX P2000_IP_Business_Phone.html We have set up a forum to discuss Grandstream products which will be monitored by our technical staff as well as Grandstream: http://www.thevoipconnection.com/forums/index.php?board=4.0 All members of this list are of course welcome. Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Anyone have a GXP-2000 working with Asterisk yet?
We have had these for a while now and they work great with Asterisk. As Brian said, setup is a breeze. We have not experienced any of the audio issues he describes. They ship with an early (somewhat limited) firmware version so an immediate upgrade is probably in order (set tftp server to 168.075.215.189). Obviously, it may take a few versions to get everything perfect, but it's pretty darn stable for a new product. It has all the call control features you'd expect from a business phone plus it has a backlit display AND built-in Power over Ethernet AND 10/100 switch AND 4 independent registrations AND 7 programmable speed-dial buttons. One strange thing - the headset jack is 3.5mm and it's pretty hard to find headsets with that size jack. We did find some adapters to convert to the usual 2.5mm. US$114.95 at https://www.thevoipconnection.com/store/catalog/product_16233_Grandstream_GX P2000_IP_Business_Phone.html We have set up a forum to discuss Grandstream products which will be monitored by our technical staff as well as Grandstream: http://www.thevoipconnection.com/forums/index.php?board=4.0 All members of this list are of course welcome. Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
