Displaying 16 results from an estimated 16 matches for "zip2s".
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zip2
2004 Aug 04
0
Zultys 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
calls into the phone. Occaisionally I get a
2004 Jun 01
0
Message light and paging on Zultys ZIP2, Uniden UIP200 time offset
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
2004 Aug 12
0
Zip2 configuration via tftp?
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
2006 May 19
2
British English voice files are ready for download
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>
2006 Feb 15
6
asterisk silence suppression?
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
2004 Jan 08
5
Dialing the Phone from OS X Address Book with AppleScript, XML-RPC, PHP and Asterisk
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
2003 Jun 27
2
Zultys SIP Phones - NEW?
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
2003 Oct 16
0
Zultys Zip 2 Registration / Disabling SIP Authorization
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
2004 Jan 20
1
G729 - how many needed?
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
2005 Feb 16
1
RTP Stream on Multicast
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
2005 Jun 15
1
SIP transfer/REFER to voicemail problem
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
2005 Sep 29
0
DTMF tones from PSTN not reaching SIP device
Greetings, I am PRIs connected to a Cisco 36xx gateway, which in turn
connects to Asterisk via SIP. The problem I am having is that DTMF tones
originated on the PSTN side are not heard on the SIP device. On the other
hand, tones originating on the PSTN side are received by Asterisk when
talking to voicemail or an autoattendant.
>From the Cisco debug, I can see the Cisco sending NTE (RFC2833)
2006 Jan 17
0
rx/txgain per device?
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...
2007 Nov 28
1
Polycom MWI's will not turn off
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 '
2012 Apr 23
0
[LLVMdev] SIV tests in LoopDependence Analysis, Sanjoy's patch
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;*
*
2012 Apr 12
6
[LLVMdev] SIV tests in LoopDependence Analysis, Sanjoy's patch
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
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