similar to: Zultys Phones w/ Encryption

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

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

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.

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

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 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

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

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 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

Hello All, I'm having trouble getting a Zultys ZIP2 to work with Asterisk, along with some other troubles in general. I keep getting a "Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from x.x.x.x). Even when Asterisk reports that the ZIP2 registered correctly, I can't make any calls out from the phone, or calls into the phone. Occaisionally I get a

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

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

I have had the opportunity to test many IP phones in the last 6 months and I thought you might enjoy a quick review of what I have found. Grandstream Budgtone 200 - Poor Quality for business use- Looks good, and the handset feels nice, buttons have a decent feel, but the disply is difficult to read when you are not directly over head of the phone, plus the sound quality of the handset, and

Anyone had experience with this phone? The interesting features that caught my eye were the RTP encryption, speaker "pager", and built in 4-port ethernet switch. Of course, RTP encryption doesn't exist in * yet, but it might be interesting (I've had people ask about it, but I don't know how serious they are about needing it.) http://www.zip4x4.com/summary_ZIP4x4.htm

I just got a flyer from my buddy on these phones today, totally SIP based, includes the G.729 speech compression codec. http://dm.zipphones.com/dm/zip2/index.htm Any word on these? -- Mark Street, D.C. Red Hat Certified Engineer Cert# 807302251406074 -- Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 GPG key http://www.streetchiro.com/pubkey.asc

Hi, I looking forward to integrate Rail3 application with SourceForce. Any help in this regard is appreciated. Thanks, Balakrishna -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFF+G/Ez6ZCGd0@public.gmane.org To unsubscribe from this group, send

Pulled Stable. ( CVS-v1-0-01/12/05-11:43:32) This does compile cleanly. Now I just have to weed out the features from CVS-HEAD that I was using and patch them into STABLE. -brian On Wed, 12 Jan 2005 at 11:37 Paul Rodan (asterisk@glitch.cc) wrote: > Problems with CVS-HEAD, but what about CVS-STABLE? > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com

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

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

Hello, I am trying a setup that is the following: SIP Phone (Zultys) --> Asterisk ---> H.323 GK (Cisco) ----> PSTN Any calls from H.323 GW through GK goes to PSTN, no problem. SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem. SIP Phone to PSTN, rings normally, on the PSTN, then connects when the PSTN phone picks up, no audio on both directions. PSTN GW support