similar to: Snom Phones Volume

Hi All, Does anyone on-list have experience with the Zultys 4x4 and 4x5 phones with AES encryption? Does this work well with *? I imagine that it would work as long as * doesn't have to decode the stream. What are the implications for calls going to voicemail or MeetMe? I'd pondering installing the 4x5, especially if Zultys can make the on-board FXO port available to * as well as a

Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

Hi All, A friend has asked me to get a new Zultys 4x5 working with his * server. I've been over the web gui but don't see where all the registration info gets entered in the phone. Is there someone on-list who has Zultys phones and can advise me as to their setup? The manufacturers docs are useless. Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr.

Does anyone on-list have any advice about the version 2.2 firmware for the Zultys 4x5 phone? It has a new "gateway" mode that is supposed to direct calls on the analogue line to the PBX for VM, etc. Zultys has not yet presented any docs, and the keep refering me to the reseller who doesn't know anything about t at all. Thanks, Michael -- Michael Graves

Hi peoples Can anyone tell me if the Zultys Zip 4x5 supports iax protocols or if they have configured one before for iax. If you have a sample config file that would be great. Any assistance would be nice Scott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/6c243053/attachment.htm

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

I'm trying to get a Zultys Zip 2 phone working with Asterisk. The phone seems to be failing registration (see sip debug output below). However, I can place calls TO the Zip2 from other SIP phones (Grandstream BT-101, Xten X-Lite, and eStara Softphone) and from Nortel PBX extensions coming in to Asterisk over a PRI T1. The problem is that I cannot dial any extensions from the Zip 2. Any

Hi, I'm trying to setup virtual mailboxes. My setup: - migrated from Courier, so: separator = . prefix = INBOX. inbox = yes In doc i have read, thant i need to create separate namespace. namespace { prefix = virtual/ separator = / location = virtual:~/Maildir/virtual So i need to change / with . (Dot) in prefix and separator? Do i need to append LAYOUT=maildir++ to location? Where

Good Day List, I have my asterisk box setup to be an ntp server, and my zultys 4X4 phone is able to get the time, however I must first select the TimeZone Offset and then it will use the time setting from my server. This is a hassle because every time the phone reboots the user must answer this question and as you can imagine End users do not know what to do and since the phone is not

I am trying to get a new Asterisk installation running using a Zultys ZIP2 phone and a Uniden 200 phone. I have the system working reasonably well (although probably not optimal) except for a couple of items. First, I can't get the voice mail message light to work on the Zultys phone but it works just fine on the Uniden phone. Second, the time presented on the Uniden UIP200 phone is 1 hour

I've got a Zultys WIP2 and Zultys 2x2 both of which support encryption. I have patched my asterisk with srtp (srtp.sourceforce.net) as well as with the patch found at http://bugs.digium.com/view.php?id=5413. I'm trying to utilizing the encryption feature of the two Zultys phones to create an encrypted call, but am having extreme difficulty. I keep getting the following error message

Hi everyone, I just installed asterisk on my system with the purpose of rerouting calls on sip channels. I don't think i need any hardware for that. I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both of them and to call one from the other on the same machine, however could not. I 1-) I could connect sjphone in isolation to freeworld dialup howver i got no sounds

Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed for '204.194.36.138' The telephone LCD says "SIP registation

Hi all, I have found the solution for getting Zultys handsets to accept paging messages. We need an app to send RTP packets with a codec of G711U to send to port 3771 on multicast dest of 224.0.0.1. There isn't any need to use SDP or SCCP packets to inform the phone to "listen" for those packets, they do it by default. Finally we may be able to get a solution going for Zultys

I have my first Zultys here but can't seem to get to first base. Does anyone have a basic configuration file they might share? Mine is a Zip 2+. If I understand the Zultys User Manual, the default passwd should be a null string--but the web interface isn't accepting that (nor 'admin' nor a couple others). Anyone know what the default passwd actually is? Thanks for any and all

Hey Cats, I''m having a problem. I''m working with an object that contains a bunch of arrays: var myObject = { "data": [ {"line": [1,"aoo","far"]}, {"line": [2,"boo","ear"]}, {"line": [3,"coo","dar"]}, {"line":

We just did an upgrade on our Asterisk to the CVS version and our Mediatrix 1102s stopped working correctly. Our Asterisk is connected to the PSTN with a PRI. Calls from the PSTN to the Mediatrix 1102 work fine. The issue is calling out to the PSTN from the 1102. Asterisk looks like it process the call just fine except there is no talk path. Get this, though: If you flash hook and then

When installing xen-utils-V in a driver domain domU, it drags in xen-utils-common, which also contains the init script for xenstored and xenconsoled. Installing the package will fail right away, because it exits non-zero after checking whether it's running in a xen dom0 or not: systemd[1]: Starting LSB: Xen daemons... xen[7215]: Starting Xen daemons: (warning). systemd[1]: xen.service:

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>