similar to: Zultys ZIP2

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'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 would like to configure my Zip2 phones via tftp, however the tokens in the config file are (apparently) not all documented. Specifically, the username/password/callerid fields seem to be only configurable via the web interface. I find this hard to believe, but the documentation and examples that Zultys provides don't help. If you have an example of a tftp-loadable config file for the Zip2

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 all, Does anyone know of a method of sending a raw G711 stream to an address in Asterisk. For example, an application that takes a argument of a phone and a port. The reason? I have found a method to paging on Zultys ZIP2 and ZIP4x4 handsets. Basically it involves sending a stream of RTP data to port 3771 to multicast address 224.0.0.1. Would it need to involve me writing my

I have purchased a single G729 license - however, how many are actually needed? All my IP phones have G729a codecs built in (Cisco 7960 / Zultys ZIP2) - I would have assumed that if the phones can do it, and canreinvite=yes, then the phones shouldn't need to go through asterisk anyway? For calls that do go through asterisk, is a single license required for each side of the stream? (i.e. a

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 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 all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

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

Is it possible to adjust the rx/txgain values per device? I have a mishmash of different phones (soft phones & sip hard phones) and each of them sound different w/the same rx/txgain settings. Is there any way to adjust these via asterisk? I'm having the most difficulty with sjphone & these zultys zip2 phones, vol is right on one, but way off on the other. Thx again...

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

Dear all, I have spent some time tying to get the system attendant patch to work; http://bugs.digium.com/bug_view_page.php?bug_id=0000214 I get no errors patching the system and the function runs, but I keep getting the following error; queue: Nexus1, options: (null), url: (null), announce: (null), timeout: 0 -- Started music on hold, class 'default', on SIP/phone10-a3f0 --