Displaying 20 results from an estimated 2000 matches similar to: "Stupid Question #1 - Testing the "s" exten from a SIP Phone"
2005 Oct 17
6
initiate call recording from phone.
I am looking for a way to allow a user to record a call simply by
pressing a button or some combination of buttons on their phone. Is
this possible?
I have read the stuff about the monitor command; however, this doesn't
seem to be very interactive for the user.
Thanks,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
2007 Oct 26
2
Initial review of American Telecom X10001P DECT/SIP phone
Mojo with Horan & Company, LLC wrote:
> And it makes *clear* calls assuming you're within allowable range.
> Speakerphone seems to work well too.
I meant to mention that the DTMF tones and dialtone sound like they're
played at such a high volume that they clip through the handset's
speaker. DTMF is rfc2833, so what I'm hearing through the handset isn't
affecting
2006 Mar 13
2
Simple php script to monitor asterisk calls
Hiya, hope I don't bore anybody with this. There are certainly a lot of
monitor-y things out there and they just didn't fit my need, so maybe
this will fit someone's besides mine.
http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one
is a php script called pbxmonitor, and one is a flat file of extensions
to extension name mappings of internal users. It
2006 Jan 25
4
Setting ringtone on Polycoms
Hi,
I'm having trouble setting the ringtone on my Polycom 501.
The relevant entry in extensions.conf is:
exten => 801,hint,SIP/creative1
exten => 801,1,SetVar(ALERT_INFO="Test")
exten => 801,2,Dial(SIP/creative1,20,Ttr)
In the sip.cfg:
<alertInfo voIpProt.SIP.alertInfo.1.value="Test"
voIpProt.SIP.alertInfo.1.class="13"/>
and
<TEST
2007 Oct 26
2
Checking Multiple VM?
Hi,
On my asterisk I have a voicemail for my extension and my Soft SIP phone
tells me when there is mail for me.
I also have other voicemail boxes which are not tied to any specific
extension - rather they are for incoming callers when they interact with
an AA, such as "to register for the event please press 1 and leave your
details"...
Can I chain or aggregate voicemail boxes so
2007 Sep 05
7
Can asterisk give half-ring periodically for MWI?
Hi all,
Configuration: Analog phone connected to TDM400p.
I'd like the phone to give a half-ring (chirp) periodically when there
is a message waiting. Can this be done? How is it configured?
The visible "Message waiting" indicator and the stutter dial tone are
working fine, but are not sufficient for me.
Thanks!
2008 May 01
1
Remote host can't match request NOTIFY???
Hi all,
I'm seeing a lot of these messages:
[Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response:
Remote host can't match request NOTIFY to call
'2e4a02807750b7024d5ff09c668fa0f7 at 10.0.0.2'. Giving up.
[Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response:
Remote host can't match request NOTIFY to call
'0755ad8f40b9d09d491b635e70bb8905 at
2008 Apr 06
3
Need help with Cisco 7960
Hello all,
I need some help with my Cisco 7960 enabling TFTP. Does anyone know what numbers to press in the menu? Or can I enable this through telnet?
Many thanks,
Christian
2007 May 15
1
Astsee v0.1 released - an Asterisk channel monitor for linux/X windows
Hiya everyone. I have been working on a fun little app to watch what's
going on in your asterisk box via its manager interface. There's a
screenshot up and some info at http://sitkavoip.com/astsee/ -- Sorry it
requires allegro, but I was more keen about getting the ideas down than
worrying about the framework.
Comments/questions welcome, but probably off-list is best unless they
2007 Aug 23
3
Asterisk Prompt
Hi List;
I read the following sentence:
"The CLI prompt is set with the ASTERISK_PROMPT UNIX
environment variable"
In the following link:
http://www.voip-info.org/wiki/index.php
page=Asterisk+CLI+prompt
The question is: what is the ASTERISK_PROMPT UNIX
environment variable and where I can access it to
change it? Also where I can find information about it?
Regards
Bilal Ghayad
2005 Dec 28
3
voip-info: Asterisk record calls
On this page http://www.voip-info.org/wiki-Asterisk+record+calls there
is "Example by Mojo". I have done everything he said and I have sox
package installed.
[root@pbx recordings]# sox -help
sox: Version 12.17.7
...
When I open this web page http://10.0.0.26/recordings/index.php I get
this: No Recordings Found
And there are recordings in /var/spool/asterisk/monitor
Do I have to do
2005 Oct 05
4
dropped calls when g729 is used on sip leg
Hello - I have 8 polycom 501s all setup great using ulaw. We have put
them through a pretty rigorous torture over the last 4 months, and
they've performed famously. No dropped calls ever.
We invested in some g729 licenses. changed my ipmid.cfg so that g729 is
priority 1 and ulaw is priority 2. I added allow=g729 to my extension's
sip.conf entry, where existed before disallow=all
2005 Oct 12
2
Polycom: Button Remapping, HELP!
I need to find a way to have the Polycom phones automatically park
calls. Right now my users hit #70# (I know the last # is optional but
it speeds it up.) to park a call. Personally I think this is easy, but
my users would like one button to do this for them. The Polycom has
buttons we don't need (Transfer & Services), it would be nice if I could
remap one of those buttons to dial
2009 Apr 16
1
AGI Programming
Hi all,
This isn't meant to be spam I thought some of you might find it interesting.
Packt Publishing approached me a few weeks ago and asked if I would like
to review a book or two for them on my blog.
The first one they sent me is called Asterisk Gateway Interface
Programming and has only just been released. It was written by Nir
Simionovich.
You can read my review here:
2007 Nov 28
5
To DB or not to DB?
I lurk and comment a little on here and have been playing with * for a
short while.
I am interested in hearing about the pros and cons for using a database
backend to Asterisk. My current setup is simple, out of the box with
config files in /etc/asterisk and logs etc going into /var.
I notice a great many of the contributors here seem to use a db backend
(is this also called Real Time
2008 Apr 15
2
dialed number notify at invalid dial situation
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised [mainmanu].
I wish to notify invaid command like a following
exten => i,1,playback('your command is ...')
exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-(
exten => i,3,playback(' is incorrect! please again ')
# This exten lines are figure for instruction.
# I
2007 Jun 06
2
Console duplicate output problem
This is really strange. Every message to the (VGA) console is written
twice to the screen, but not on the SSH connection.
Any clues how to stop this behavior?
-- Executing BackGround("Zap/216-1", "custom/3566/91_0000|m|") in
new stack
-- Executing BackGround("Zap/216-1", "custom/3566/91_0000|m|") in
new stack
Bart
2008 Mar 26
5
Asterisk parking hold and transferdigittimeo ut
> -----Urspr?ngliche Nachricht-----
> Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com]
> Gesendet: Dienstag, 25. M?rz 2008 23:23
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: Re: [asterisk-users] Asterisk parking hold and
> transferdigittimeout
>
> It seems that the dialplan comes into play. If your parking
>
2005 Oct 06
2
Mediatrix 1204 and Asterisk
Dear Group,
I have my Asterisk box working with a Mediatrix 1204.
I have 2 questions;
1) I do not seem to get a Call ID on the call coming via the Mediatrix
1204. I was wondering if anyone had this configured and if they could
share this with me?
2) How do you route a call based on caller ID on Asterisk. At the moment
I'm routing calls via DNIS.
Thanks and Regards
Shad Mortazavi
2005 Oct 06
2
how do I know what codec is being used
Hi,
This may be a stupid/easy question for many of you.
Q. how do I know what codec is being used for a particular call or call
leg?
Thanks.
AK
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