similar to: Hardphone deployment recommendation

Hello All, I am looking for a sip phone that is capable of automatic nat. The Cisco ata186 for example works fine for natting with iconnecthere, but as for asterisk, both my 7960 and polycom ip600 require you to set the nat ip on the tftp. Does anyone know a good phone (or ata) that can do this automatically? For example, I want to give a phone to my brother, who is going to europe. His ICH

Hi there, Do anyone have any experience with SIP phone that support a headset ? We have Budgetone phones but we need headsets. What would you advise ? Thanks Thibault

Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've

This is a wierd one. Can't figure it out. I have an SPA-3000 at the house handling my incoming line. It's setup to direct the incoming call to asterisk. Works great 99% of the time. A few times a day, I'm getting calls that ring once internally and are then hungup. I managed to get a detailed log [1] of what's happening today and it looks to me that the SPA is acting wierd.

After working with the Grandstream GXP 2000 series phones, I have decided that I am quite unhappy with their problems, both voice quality, volume, features and others. For their price now, there are plenty of phones to choose from as well. So subjectively what would be the best Hardphone for a small/medium business with multiple line support, BLF, etc. Are the Cisco 7960 the best of the

Hi, I'm considering giving the Grandstream BudgeTone-102 phones a try. I've been using Cisco 7960's to date, but the low cost of the Grandstream phones are hard to ignore. I have two questions: 1) Does the message waiting indicator on the BudgeTone's work with Asterisk? 2) The one line 12-digit LDC concerns me a bit. Is the LCD able to display both the CID number and name on

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?

Hello, How good is :SPA 841 form SIPURA. Thanks Varun

> Just wanted to let you know you were just left a 2236:45 long > message (number 1) in mailbox 1000 from an unknown caller, on Monday, > February 21, 2005 at 05:38:14 AM so you might want to check it when you > get a chance. Thanks! > > --Asterisk Hmmm... Call came in either via a SPA-3000 or an X100P Zaptel FXO interface. Got 2200 minutes of beep-beep-beep-...; the caller

I have a SPA-3000 that is failing to hanging up pretty often; almost every day now. The weird thing is that an almost identically configured (same FW, different HW rev) second unit right next to it isn't having the same problem. Swap the lines and the problem stays with the unit. I've been going round and round with Sipura and their latest message told me to use a procedure detailed in

This is probably going to sound really silly and I must be confused about it. Maybe someone can set me straight. I've been tinkering for a while with * and a number of different FXO/FXS cards, SIP phones, and ATAs trying to get a feel for what works and what doesn't. In the SIP phone group, I have a Cisco 7940G, a Polycom 500, and now an SPA-841. Each of these allow me to configure at

Am I doing something wrong here? GOt a SPC-841 the other day and have it registering properly. Can place and recive calls as expected but when on the phone, a second call is immediately dumped to "busy" voicemail. Does this thing not support call-waiting? Or, have I just got my configs wrong? Paul -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com

Running * CVS with a SPA-841 (0.9.5) and can't seem to get the message waiting light to come on automatically. There is a control in the web interface to turn it on and off (seems rather curious to me but whatever). Has anybody got an idea as to where I may be going awry? TIA, Paul -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com 1711 Indian

I have an asterisk Redhat 9 box running 4 hardphone extensions. Inter-extension calls are crystal clear. However when I dial out through my iconnect account I get a lot of jitter. At first I thought it was my nat gateway but after I programmed on of the hardphones (budge tone 100) for direct dial to iconnect I have clear voice transmission. I have no way of explaining this. My asterisk sip.conf

I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad.

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

G'Day All, How do I fix this: I receive a call at the extension. Press the hold button. Music on hold starts. When I place the handset back on the cradle, the call gets hung up/disconnected. The Phone is A GrandStream Budge Tone 100. Thanks