search for: optipoints

Hi there, I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I?m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I try to call one of the optipoints from a Sjphone, or from one opti to another, then no connection or sign...

Hi everybody, I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk. It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers. The Optipoint shows "no Server..." (Registrar?) in Display. Sip debug shows no unusual (to me) Messages. Sip show

Hello, I?m looking for Optipoint 420 Standard SIP Firmware to make my first tests with Asterisk and SIP, but I?m unable to find it. Could someone help me? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070415/c2da9cc0/attachment.htm

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

Anybody using Siemens optiPoint 300 H.323 phones? I saw a few references to them in the archives of this list, and the Wiki seems to be down. I have a chance to pick up a bunch of these, cheap. Questions: * Asterisk support? * What sort of power supplies will they need? The bunch I am looking at are surplus and have no supplies. Thanks, </edg> Ed Greenberg San Jose, CA

Not strictly a problem with Asterisk but one of my phones. A couple of days ago I decided to update the firmware in my Optipoint 600 Office which looked as though it went swimmingly until that is, it rebooted. Since then the phone just boots up and displays the following: Can't Boot!! Development shell active. It doesn't try to request a DHCP address, in fact it does seem to do

Hi all, I have an optipoint 400 Sip phone with 2.46 sip image which stores its music on hold file localy (on the phone) rather than indicate to the server the call is on hold. i'd like to have Moh play from the server, and i'm not sure if there is a work around to achieve this anyone else has the same phone and came across similar problems ?? thanks fam

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been

Dear, Is there somebody who have experience with the Siemens Optipoint 400 standard SIP ? And where can we buy it (i'm from belgium) We are using for the moment Cisco 7960, 7905, snom 200, Mitel 5055 and in my opinion the Mitel does his work the best combined with the 7905, the 7960 is realy anoying to transfer, On the snom 200 you can't dial a number veryfast due to the

I have not seen anybody on the web to have found this so I thought I would check here. Anybody got this firmware? I've found firmware for the 400, but it doesn't seem to load in the 410.

Hi all, I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the softclient. Any ideas? Thx! sip.conf [2005] type=friend callerid="OptiPoint" <2005>

Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm

Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads

Hi, I'm using asterisk 1.2.1 on a debian sarge distro. I've followed notes in http://www.voip-info.org/wiki/view/Polycom+auto-answer+config and http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO but I still cannot change ring style via asterisk using exten => 666,1,SipAddHeader(ALERT_INFO="ring3") in extensions.conf . Is it

Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

>When I call a SIP user, the phone should ring in more than one >extentions. Also more than one phone should be able to register with >asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for

Hello, Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone. Thanks -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have an optipoint 400 Sip phone with 2.46 sip image which stores its music on hold file localy (on the phone) rather than indicate to the server the call is on hold. i'd like to have Moh play from the server, and i'm not sure if there is a work around to achieve this anyone else has the same phone and came across similar problems ?? thanks fam

is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working????? I am using chan_h323 Thanks!! Sebastian Nocetti. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date:

Greetings, I purchased a WiSIP at the VON conference and am now trying to configure it to work with Asterisk. I have read all of the previous postings regarding the WiSIP and most of the information apparently does not apply to the version of firmware installed on my phone (version WF.00.0F). I cannot get the WiSIP to register with my Asterisk box. It leases an IP from my DHCP server, then