Displaying 20 results from an estimated 1000 matches similar to: "web app to playback recorded phone calls."
2008 Apr 03
12
Web page to show online extensions?
Hello
Has someone written a web page (preferably PHP) that simply shows what
extensions are currently online?
Thank you.
2008 Mar 13
2
SNOM on "Do Not Call" list????
Some light relief ....
SNOM say "Please note that you will not be able to reach us by phone."
http://www.theregister.co.uk/2008/03/13/dont_call_us/
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com
2008 Aug 21
2
Siemens Gigaset IP in USA (S685 IP in particular)
For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP
range in the U.S. I'm particularly interested in the Gigaset S685 IP.
Since it's DECT 6.0, and there's an English (UK) version, I'm thinking
it should work just fine, after dealing with the walwart issue (and
maybe caller ID signalling).
Anyone imported one from the UK and using it in the US? for how
2005 Nov 02
2
Bug report on get.hist.quote
> get.hist.quote(instrument="INR/USD", provider="oanda", start="2005-10-20")
trying URL 'http://www.oanda.com/convert/fxhistory?lang=en&date1=10%2F20%2F2005&date=11%2F01%2F2005&date_fmt=us&exch=INR&exch2=&expr=USD&expr2=&margin_fixed=0&&SUBMIT=Get+Table&format=ASCII&redirected=1'
Content type
2008 Jun 06
1
Asterisk not picking up incoming calls from TDM400P
Hi,
I am having some issues with a new server install in Singapore.
Outbound calls work fine.
Inbound calls are not picked up by Asterisk.
Zaptel 1.2.25 and Asterisk 1.2.28 both built from source.
libpri installed
wctdm and zaptel load without error
Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface
Registered on major 196
Jun 6 23:34:03 fs01 kernel: [211138.372937]
2007 Jun 12
2
Transfer caller direct to voicemail
Hi,
Our operator frequently gets requests to transfer a call directly to
voicemail in order for the caller to leave a message without disturbing
the callee. Basicly, I'm looking for a blindxfer to vm.
My first thought was to prepend a digit (eg 7) to the extension but this
does not fit well with our dialplan.
According to an article on voip-info.org Asterisk@Home appears to
implement
2007 Mar 28
3
Multi-line phones - Asterisk uses wrong callerid
I have some phones (and an ATA) that are shared between two users who
each have separate voicemail but they are not behaving as desired nor
expected.
Incoming calls show up on the correct lines.
Calls originating from the device are seen, at the terminating device,
as coming from the account listed last in sip.conf, regardless of the
line selected.
This creates three main issues I would like
2008 Aug 15
1
Problem with Aastra 480ci and qualify=yes
Hi,
We have a few Aastra 480ci phones and we've noticed that in order to
get the phone to receive a call, qualify must be = no.
Apparently the Aastras do not respond to the qualify message (or
respond in a way Asterisk doesn't understand) and Asterisk thinks the
phone is unreachable.
However, this now prevents MWI from working properly on the phones.
Does anyone know how to get MWI
2011 Apr 09
1
How do I make this faster?
I was on vacation the last week and wrote some code to run a 500-day
correlation between the Nasdaq tracking stock (QQQ) and 191 currency pairs
for 500 days. The initial run took 9 hours(!) and I'd like to make it
faster. So, I'm including my code below, in hopes that somebody will be able
to figure out how to make it faster, either through parallelisation, or by
making changes. I've
2007 Mar 26
9
Multi-registration ?
Hello,
1. Is it possible to install several SIP softphones on the same PC, have
them registered to the same Asterisk server and attribute to each softphone
a specific extension, ringtones or call forwarding rules ?
2. Is possible to do the same with SIP hardphones ?
Regards
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2008 Sep 02
1
R Newbie: quantmod and zoo: Warning in rbind.zoo(...) : column names differ
Hello;
I am trying following but getting a warning message : Warning in
rbind.zoo(...) : column names differ, no matter whatever I do.
Also I do not want to specify column names manually, since I am just
writing a wrapper function around getSymbols to get chunks of data
from various sources - oanda, dividends etc.
I tried giving col.names = T/F, header = T/F and skip = 1 but no help.
I think
2006 Oct 30
6
How to do Automatic Daylight Saving on Grandstream GXP-2000
Hi,
I'd set the daylight saving option to yes on all the GXP-2000 phones, but
apparantly it doesn't move it an hour back on last sunday of October. So now
I am stuck will all the phones showing the wrong time. Isn't there an option
so that it'll automatically update daylight savings?
Thanks
--
Zeeshan A Zakaria
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2007 Jul 26
1
Grandstream RTP keepalive packets causing Asterisk warning
Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where
the phone did not send rtp keepalives when on mute (resulting in
disconnect from tech support hold and concalls)
A side effect seems to be that Asterisk pops the following warning on
the console...
Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short
Grandstream say they are not sure what it is but
2008 Feb 05
4
How to hookup to cell phone for outbound calls?
Hi
I need a small PBX for use on the move. This means that outbound calls
will need to be made over the cell phone network.
Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot
then what hardware options do I have to get an outbound cellular
channel? Options need to be rock solid, so no bluetooth to a cell phone
kind of solutions need apply.
Can any of the 3G usb
2008 Jan 31
7
pulling my hair out over voicemail
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last week I got
everything setup, calls were working, etc.,.
I cam across a tutorial for voicemail, followed it, and it worked. When
I call my phone and dont answer, it goes
2008 Jul 27
3
OT - How to test tftp for phones provisioning
Hi,
I don't understand why a SIP hardphone can't provision itself using tftp.
I'm very suspicious about my tftp daemon but I lack basic knowledge of Linux
CLI to pinpoint what's going wrong and separate what belongs to SIP phone
configuration from what comes from tftp server.
What I would like to do is to add a given file in current /srv/tftp
directory and test by hand that tftpd
2006 Mar 26
2
Web based voicemail client
I'm looking for a good web based voicemail client that can use mysql or
realtime drivers. I can't seem to get vmail.cgi to work with realtime.
Thanks for any help you can give.
2007 Mar 07
1
auto dialer
Not able to get the auto dialer part of asterisk to work with the zap
channel. It works great with the sip channel. Here is the call file and
the CLI output
Call File
Channel: ZAP/G1/6144994925
MaxRetries: 3
RetryTime: 40
WaitTime: 2
Context: amaxx
Extension: 36652
Priority: 1
CLI Output
Connected to Asterisk SVN-branch-1.4-r57207 currently running on
VoIP-PBX (pid
2007 Mar 07
1
Problem HandyTone 488 does not call transfer
Hi
I have a analog phone connected to my Gateway Handytone and registered to
Asterisk 1.4 I have configured my HandyTone 488
(in the section FXS Port) for make and receive calls, however I can
not transfer a call when it come via PSTN. But, when a call come from via IP
I can transfer it.
[phoneanalog]
type=friend
secret=XXXXXXX
context=local
nat=no
qualify=yes
host=dynamic
dtmfmode=rfc2833
2007 May 08
1
Problems witch SPA3102.
Hello,
i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with
cdr. Well all I want is to receive incoming calls from pstn on specified sip
account (suppose 8000), and to initiate outgoing calls from all my asterisk
sip accounts through SPA3102 device. Someone can explain me what may i set
on SPA and asterisk to do this thing. Thank you for your support.
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