search for: zip2

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 cal...

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 Unid...

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....

Hi folks, With thanks to Alison Keenan (another Alison!) for the voice, Chris Bagnal for converting from 44k wav to sln and finally Terje Elde for debugging my HTML code, the British English files are now ready for download. They can be got from http://www.enicomms.com/cutglassivr/ Thanks and don't forget to practice safe IAX ;-} Mark -- Mark Phillips <g7ltt@g7ltt.com>

...e 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 to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads? Thx as always

I run an Apple OS X workstation and I've got a server on the same LAN that's both a webserver and an Asterisk PBX. I wanted to be able to originate calls in the OS X Address Book application, and have Asterisk dial them and connect them to the phone on my desk. I've assembled a system that uses AppleScript to connect, via XML-RPC, to a web application that, in turn, connects to

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

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 extension gets an immediate busy signal and NO output from sip debug (not even any...

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 connection between two phones needs two licenses?)...

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 own application for asterisk? Thanks Mathew -------------- next part -------------- An HTML attachment was scrubbed... URL: ht...

...he voicemail prompts? Would Asterisk be doing a half duplex protocol conversion? Any insight would be greatly appreciated!! Current configuration: Fedora Core 1 Asterisk - 1.0.7 (had same problem on 1.0.6) SJPhone - 1.50.271d, Mar 11 2005 (WinXP) XLite - 1103m build stamp 14262 (WinXP) Zultys Zip2 - ZUTS 3.52 sip.conf exerpt: [6003] ; (A) type=friend regexten=6003 username=6003 host=dynamic disallow=all ;allow=gsm allow=ulaw [6004] ; (C) type=friend regexten=6004 username=6004 host=dynamic disallow=all ;allow=gsm allow=ulaw [2101] ; (B) type=friend regexten=2101 username=2101 host...

...can see the Cisco sending NTE (RFC2833) RTP packets to Asterisk and it appears that Asterisk is propogating them down to the SIP device. However when tones are pressed on the PSTN side, all that can be heard on the IP phone is silence. I've tried this on three different IP phones (Cisco 79xx, ZIP2 and Sipura) and they all behave the same, leading me to conclude it isn't a phone config problem. Everything (Cisco and phones) are configured for dtmfmode=rfc2833. Anyone got any ideas? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

...ible 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...

Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog '

Hi, When I write various test cases and explore how they're handled by the code in LoopDependenceAnalysis::analysePair, I'm surprised. This loop collects pairs of subscripts from the source and destination refs. * // Collect GEP operand pairs (FIXME: use GetGEPOperands from BasicAA), adding* * // trailing zeroes to the smaller GEP, if needed.* * GEPOpdsTy destOpds, srcOpds;* *

Hi, Here is a preliminary (monolithic) version you can comment on. This is still buggy, however, and I'll be testing for and fixing bugs over the next few days. I've used your version of the strong siv test. Thanks! -- Sanjoy Das. http://playingwithpointers.com -------------- next part -------------- A non-text attachment was scrubbed... Name: patch.diff Type: application/octet-stream