A very nice feature of another conferencing system that I've used is
that the admin/moderator can press a star code to MUTE ALL OTHER USERS
on the conference.
This is great if you have several people on the call and one of the
people puts the call on hold (and so the music/advertisement/your call
On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and
6730i, but none of them indicate the voic-email. Where should I look for
trouble to find the root issue for MWI?
(1) Check from the
Happy Friday everyone,
Is there a way to log the negotiated codec that was used for each call
in CDR or in a separate log file?
This is for SIP-based calls, if that matters.
Perhaps there is some variable that can be queried as part of the
dialing script;
Or is it possible to grab the codec
GoogleTalk connects ok to Asterisk 1.6.2.7 but how can you choose
voice menu options (press 1 for Bob, press 2 for Betty, ...) from the
GT client?
(There is no dial pad in the Windows GT client, but what you type in
the message box does show up on the console as an incoming Jabber
message.)
Is
I want to set up a conference call to be recorded automatically, so
I'd like the recording to start when the second caller joins the
conference (one caller already there). The recording would continue
until the last user hangs up.
How can you determine how many are already in the conference
Is it possible to use an Asterisk feature code to transfer a call to a
specific extension?
For instance, if you take a call, and the caller wants to go to a
conference, it would be nice to use a feature code for this, rather
than going through a longer transfer sequence.
e.g.:
- You have a
When you press *1 in Asterisk (1.6.2.7) to start/stop call recording,
the console CLI shows:
User hit '*1' to record call. filename: wav,auto-1273791789-103-5551212,m
Is it possible to play a sound to back to the person who pressed *1 to
indicate to them that recording has actually started or
Hello asteriskers,
I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI
prompt, and found references on using the command soft hangup
SIP/channel, but as you can see below, the soft hangup command
does not seem to exist, and there is no mention about it in the
UPGRADE*.txt
On Tue, Apr 20, 2010 at 10:49 AM, Steve Edwards
asterisk@sedwards.com wrote:
On Tue, 20 Apr 2010, Tilghman Lesher wrote:
On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote:
I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI
prompt, and found references on using
Google is your friend. You should use it. Search for:
asterisk extensions.conf dictate
or
asterisk extensions.conf dictate example
Some results:
http://www.asteriskguru.com/tutorials/dictate.html
and
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dictate
On Wed, Dec 30, 2009 at
Try putting the wait before the Answer.
...
exten = s,n,Wait(10)
exten = s,n,Answer
...
On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither smit...@c-c-i.com wrote:
Dear All,
I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by
Vitelity. When the number is called it goes to my
by the provider.
If wait ... answer doesn't work for you, you'll have to provide more
output from the CLI and tell us more about your configuration.
On Fri, Dec 18, 2009 at 10:29 PM, Bob Smither smit...@c-c-i.com wrote:
On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote:
Try putting
How about:
exten = 977,1,ExecIf($[${CALLERID(num)} =
733025975]?Set(CALLERID(num)=0317998975))
exten = 977,n,ExecIf($[${CALLERID(num)} = 1234]?Set(CALLERID(num)=317998977))
exten = 977,n,ExecIf($[${CALLERID(num)} = 5678]?Set(CALLERID(num)=317998978))
[..]
exten = 977,n,Dial(SIP/0317998977)
On
Of course, as long as your endpoints support it. Read more about it
and purchase G.729 channel licenses for Asterisk from Digium:
http://www.digium.com/en/products/g729codec.php
Once you have the codec properly installed, enable it for your peer in
your iax.conf file allow=g729. Restart
Hi all,
What's the best method to send automon call recordings (*1) to the
voicemail box of the Asterisk user?
Do you have to trap hangups, etc, or is there some global variable
that can be set?
Thanks!
S.
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, 2009 at 2:58 PM, Doug Lytle supp...@drdos.info wrote:
Steve Johnson wrote:
Hi all,
What's the best method to send automon call recordings (*1) to the
voicemail box of the Asterisk user?
I've picked up the following off the list a while ago. Works pretty
good. I do a mysql lookup to see
My Dial() command is Dial($LOCAL_DIAL)
Perhaps you should be using:
Dial(${LOCAL_DIAL})
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To UNSUBSCRIBE or update options visit:
Hello list,
How can you obtain the default mailbox for a SIP extension (as stored
in sip.conf and shown with sip show peer ext)? Is there a
function to extract it?
Why? Some extensions have shared mailboxes and others do not and I
don't want to duplicate logic, just use the extension's default
For long distances, a wireless point-to-point might be more economical
than trenching.
e.g: Carlson Wideband CDMA Spread Spectrum Phone Line Extender
http://www.oksolar.com/communications/phone_line_ext.htm
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They are 2-letter ISO country codes.
http://www.iso.org/iso/english_country_names_and_code_elements
On Wed, May 6, 2009 at 1:05 PM, Steve Edwards asterisk@sedwards.com wrote:
I've googled for way too long, where are the 2 letter language values
defined?
I know:
en = English
es =
Also check out:
http://www.w3.org/International/questions/qa-lang-2or3.en.php
On Wed, May 6, 2009 at 1:22 PM, Steve Johnson stevej...@gmail.com wrote:
They are 2-letter ISO country codes.
http://www.iso.org/iso/english_country_names_and_code_elements
On Wed, May 6, 2009 at 1:05 PM, Steve
Anyone want to talk briefly about one-button call parking/pickup on
Asterisk with Polycom phones? Does anyone use it or know to do it?
On many phone systems there are 2 or 3 park buttons, and you can park
a call onto an unlit park button, and then the light flashes. You can
go to any other
Software:
dahdi-linux-complete-2.1.0.3+2.1.0.2.tar.gz
asterisk-1.6.1-rc1.tar.gz
Hardware:
4-port fxs card
Example:
# /etc/init.d/dahdi status
### Span 1: WRTDM/0 wrtdm Board 1 (MASTER)
1 FXSFXSKS (In use)
2 FXSFXSKS (In use)
3 FXSFXSKS (In use)
Hello all,
I have one Asterisk 1.4.21 system connected to a North American POTS
line. Normally hangup detection works fine, and Asterisk hangs up
properly if you are talking to a caller and they hang up; but
occasionally a call comes in (typically from a US telemarketer) where
the caller hangs
automatically stopped after a silence of 10 seconds
[07:46:42] -- Zap/1-1 Playing 'auth-thankyou' (language 'en')
[07:46:43] == Auto fallthrough, channel 'Zap/1-1' status is 'NOANSWER'
[07:46:43] -- Hungup 'Zap/1-1'
On Fri, Jan 16, 2009 at 12:07 PM, Steve Johnson stevej...@gmail.com wrote:
Hello
One of these methods will work:
exten = s,n,ExecIf($[${dialplan} = NZ]|Set|NAT=0)
exten = s,n,ExecIf($[${dialplan} = NZ]|Set|INT=00)
-or-
exten = s,n,GotoIf($[${dialplan} != NZ]?not-nz)
exten = s,n,Set(NAT=0)
exten = s,n,Set(INT=00)
exten = s,n(not-nz),more_dialplan_stuff
On Mon, Dec 15, 2008
Is anyone using fixed Park buttons (some of the ones on the left side
of the screen) on a Polycom phone? Here's what I mean:
- Call is received and parked, by the user pressing an unlit park
button (e.g. 701) and the call is parked there.
- The call can be picked up at any other extension by
Hi everyone,
What creative methods are used to support dial-by-name functionality
for people who go by more than one name?
e.g.: Rebecca/Becky, Margaret/Peggy, William/Bill, Liz/Elizabeth, etc.
We'd like to use the f first name option of the Directory function,
as the particular phone system
On Fri, Mar 28, 2008 at 12:05 AM, Paul Hales [EMAIL PROTECTED] wrote:
Can't you just use the same bootrom for all your polycom phones?
PaulH
On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of
Of course *it would be nice if* the IAX2 authentication parameters
were also encrypted, so that there was no danger of a 3rd party
hijacking your connection and generating a bunch of extra charges.
S.
On Fri, Feb 15, 2008 at 11:31 AM, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tim Panton wrote:
Please post your sip.conf entry for your phone and also describe your
calling path. Are you having a problem with internal calls (e.g.: to
voicemailmain) on the same switch, or are you referring to calls to
PSTN destinations via pots/pri/sip/? Also, which versions of
Asterisk, Zaptel, linux,
I am using Polycom's SIP 2.2.0047 (the current release) and am seeing
this. It seems to occur less often with extensions reload rather
than just reload, but it would be nice to fix this.
Tx.
On Jan 22, 2008 8:30 AM, Steve Davies [EMAIL PROTECTED] wrote:
On 1/22/08, Steve Johnson [EMAIL
, Steve Johnson [EMAIL PROTECTED] wrote:
I am using Polycom's SIP 2.2.0047 (the current release) and am seeing
this. It seems to occur less often with extensions reload rather
than just reload, but it would be nice to fix this.
Tx.
On Jan 22, 2008 8:30 AM, Steve Davies [EMAIL PROTECTED] wrote
Hi list,
There are many Polycom experts on this list -- hopefully someone has a solution.
With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk
causes the Polycom 601 phones to start dumping these messages to the
CLI.
-- Incoming call: Got SIP response 500 Internal Server
Hi everyone,
How can I add the equivalent of:
exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)
in a .call file? This is to support paging to Polycom phones...
Thanks for all info!
Steve
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,NoOp(Paging Ext ${EXTEN})
exten = _1XX,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _1XX,n,Dial(SIP/${EXTEN},20,L(6))
exten = _1XX,n,Hangup
Steve Johnson wrote:
Hi everyone,
How can I add the equivalent of:
exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)
in a .call file
You might take a few ideas from this combine.sh script which works for
me. It uses the combine_wave program from
http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame
program to convert to mp3.
It converts the entire directory /var/spool/asterisk/monitor/*-in.wav
files to mp3
The user will receive email notification if you have configured the
user's email address in /etc/asterisk/voicemail.conf .
See: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf
Also check the externnotify option which lets you run an external
script when new voicemail
PM, Mike Hammett [EMAIL PROTECTED] wrote:
Does what I have in the dialplan look right or am I way off base with being
able to use that script?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Steve Johnson [EMAIL PROTECTED
This is pretty easy to suppress using the configuration files. Check:
http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
On Dec 21, 2007 11:55 AM, Ugo Bellavance [EMAIL PROTECTED] wrote:
Hi,
I have a Polycom 330 that emits a beep every 30s or so when there is a
The DNS for www.voip-info.org seems to be non-responsive. Is there a
mirror of this invaluable resource site?
Tx,
Steve
dig www.voip-info.org
;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server
; DiG 9.4.1-P1 www.voip-info.org
;; global options: printcmd
;; Got answer:
;;
Is there an Asterisk CLI command to show a list of calls in progress
(for all channels: Zap/SIP/IAX2 etc).
Restart when convenient waits until the system is idle, but is there
an obvious way of seeing what's going on at the moment?
Thanks,
Steve
___
Can anyone comment on the DSCP quality of service settings on your
Asterisk server?
The network we're setting up has data on the default VLAN, Asterisk
server and phones on VLAN 4, and we're using Polycom phones with a PC
hooked up to the phone's pass-thru port.
What iptables settings are you
Thanks, Darryl,
To clarify:
in /etc/asterisk/sip.conf you have the lines:
tos_sip=cs3; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you
have something like (this
Hi List,
We have a remote asterisk SIP phone at the cottage.
I'd like it to have minimal privileges when it first registers with
Asterisk. Ideally it should be in a restricted context. Dialing any
number would intercept the call and tell the person to log on. This
way, if the phone was stolen
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