Displaying 20 results from an estimated 34 matches for "62458".
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2458
2003 Nov 07
2
Callgroups and Pickupgroups in Console/dsp
...one... since is a very special
features that probably no one will ever use
lemme know.
Matteo.
--
Matteo Brancaleoni
Espia System Administrator
Email : mbrancaleoni@espia.it
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 911
IAX(2): guest@213.140.14.155 - ext 911
Iaxtel: 1-700-56-62458 - ext 911
2004 Jan 08
5
AbsoluteTimeout Users Messages
Hi, All
Is there a provision for "AbsoluteTimeout" application to notify
called and calling party of the reason why the call suddenly ended?
This way, the parties will be much better informed, hence they
will/should not think that
their VOIP/telco provider(s) are providing bad service.
Ta
SJ
2004 Feb 03
2
Playing announcement to called user prior toConfirmation
...39; -- play an announcement to the called party, using x as
file
see last param ...
Matteo.
--
Matteo Brancaleoni
Espia System Administrator
Email : mbrancaleoni@espia.it
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 201
IAX(2): guest@213.140.14.155 - ext 201
Iaxtel: 1-700-56-62458 - ext 201
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2003 Sep 09
0
Asterisk @ SMAU
...expo, drop me an email off line,
so will be able to meet at the expo and chat a bit ;)
Matteo.
--
Matteo Brancaleoni
Espia System Administrator
Email : mbrancaleoni@espia.it
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 911
IAX(2): guest@213.140.14.155 - ext 911
Iaxtel: 1-700-56-62458 - ext 911
2003 Sep 11
1
UK Asterisk user, please pick up the white courtesy phone
So, I have submitted my configurations as public samples, and I
should have expected this situation to arise. I changed all the
relevant "private" configuration data in my samples to obfuscate or
alter IP addresses, passwords, etc. However, I left my email address
in voicemail.conf...
Let me tell you, it took THREE messages sent by a distinctly
British-sounding gentleman leaving
2003 Nov 10
1
Jitter Buffer on chan_sip
Hi,
I would like to test chan_sip with a bigger jitter buffer. Does anybody know
where in the code this is defined? I looked through it but could not find
where.
If anybody else can find it please let me know.
Regards,
Andres
2003 Nov 12
1
IAX needs a zaptel device?
Hi All,
I'm currently running Asterisk with SIP phones and an ISDN card using
chan_capi. I've just started to use IAX (GSM codec)over the Internet and
the sound is adequate. However, there is an occasional 'glitch' in the
audio resulting in lost sound or distortion. Is the distortion because
I'm using zaprtc for timing instead of a zaptel card, or is more likely
to be due to
2003 Nov 14
1
Re: 9. Zhone zplex (Angel Gomez Garcia)
Hi
I have the last firmware for zplex, if you like i send it to you, about the
second question 24s means
24 extensions so you can configurate as you wish as fxo or fxs.
Att Yelson Vivas
2003 Nov 19
1
Play a "sound" after dialing a user...
I'd like to play a sound to a user I dial (via SIP) once
they answer play the sound and then allow me to talk to them.
The new Cisco 7960 SIP code allows to set lines to autoanswer
via the speaker phone, I'd like to play a "tone" after it rings
through and then talk...
Any thoughts on how to do this?
2003 Nov 20
1
IAX2 Ethereal Plugin initial release
Lots of people seem to want this, so I've stuck it up here:
- http://almaw.com/ethereal-iax2-plugin-0.1.zip
Note that it currently only does IAX-2. I might expand it to cope with
IAX-1 at a later date, but no promises. It's fairly basic - unzip the
file and follow the README instructions.
Regards,
Alastair
2004 Jan 09
3
Very high delay
Hi
I'm using a Teles ISDN passive card configured in modem.conf.
when i make call from my sip client (xtex x-pro) to the external world i have more than 1 second of delay and echo very.
There is some tuning to do?
The performance is better with an active ISDN card or CAPI compatible driver?
thanks
mark balester
2004 Feb 02
1
Playing announcement to called user prior to Confirmation
Hello all,
As I'm sure is pretty common, I have some extensions that dial mobile numbers
after a local timeout. I would like to prompt the caller to record their
name after the local timeout and have the recipient be able to hear the name
prior to accepting the call.
Recording the message is easy enough, so I thought about doing something like
dumping them into MeetMe after they record
2004 Feb 03
2
busy tones
Hi
When I call a phone with CAPI if the phone available I hear ringing ok
but if the phone is busy I don't hear anything at all.
Also, when I call a mobile phone and it is turned off I don't hear the
operator voice answer me telling me that the request phone is turned off
or unavailable.
Any ideas?
m
2004 Feb 03
1
Mediatrix 1102 Auth
...answers to asterisk with logon data after a "proxy auth required"
any hint?
Matteo.
--
Matteo Brancaleoni
Espia System Administrator
Email : mbrancaleoni@espia.it
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 201
IAX(2): guest@213.140.14.155 - ext 201
Iaxtel: 1-700-56-62458 - ext 201
2004 Feb 03
1
GS and NAT
Hi all.
Is it at all possible to have a GS B101 NATed with firmware 1.0.4.40?
I've tried both STUN and not STUN. The odds seems best with stun because
the phone registers with right ip adress.
When the connection is made * sends rtp packets to the right destination
AND port, but the phone doesn't accept the packets.....
Should I burn my D-LINK 604 or upgrade the GS?
/t
2004 Apr 23
1
CAPI and Extensions.conf Security problem
Hi,
I've installing a AVM Fritz Card in my ASterisk Box
I've configured everything and its running perfectly.
The problem is that everybody is allow to call through it.
Explaination:
All users registered in Asterisk can make a call towards the ISDN network
But, everybody from the Internet, knowing the extension of CAPI in the
dialplan, can call through my Asterisk to any phone
2004 May 06
3
mpg123 versions ?
We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil.
What versions does everyone use without problems.
0.59r is PERFECT
bkw
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2014 May 03
1
CONTEXT=SORT
Any reason this is not listed in the CAPABILITY string?
I see it listed as a TODO here: http://wiki2.dovecot.org/Roadmap
But it seems to work fine (as of 2.2.12):
1 UID SORT RETURN (CONTEXT COUNT) (SUBJECT) UTF-8 UNDELETED
* ESEARCH (TAG "1") UID COUNT 12
1 OK Sort completed (0.000 secs).
2 UID SORT RETURN (UPDATE COUNT) (SUBJECT) UTF-8 UNDELETED
* ESEARCH (TAG "2") UID
2003 Nov 12
1
X100P random hangups.
I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason.
I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at?
My zap config looks like.
context = inbound-work
include => extensions
signalling
2004 Jan 22
4
Gsm + snom phones
...*. How's now the situation?
the snom gsm works well with * ?
Thanks for any info, Matteo.
--
Matteo Brancaleoni
Espia System Administrator
Email : mbrancaleoni@espia.it
Web : http://www.espia.it
Phone : +39 02 70633354 - ext 201
IAX(2): guest@213.140.14.155 - ext 201
Iaxtel: 1-700-56-62458 - ext 201