Displaying 20 results from an estimated 1000 matches similar to: "Allison Smith AMA"
2010 Nov 13
2
asterisk 1.8 fax woes
I upgraded from a perfectly working 1.6.2 asterisk installation
(including fax via app_fax_digium) to 1.8.0 this evening.
All my custom modules (including swift <thanks darren!>) are working
fine except for fax.
When a caller connects, asterisk switches to the fax context and hangs
up the call.
i've captured with:
core set verbose 10
core set debug 10
fax set debug on
sip
2010 Nov 15
2
SIP calls destroyed after 1:20
After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP
calls are being destroyed after 1 minute and 20 seconds (80 seconds).
Asterisk is sending a BYE message - I have no idea why.
http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug.
can anyone suggest how i can further deal with this?
--
Jeremy Kister
http://jeremy.kister.net./
2010 Aug 09
0
Allison Smith Hilarity
Greetings and salutations Asterisk community,
I've been contacted by a man who has generously posted some prompts he
commissioned from Allison Smith. If you haven't heard Allison in humor
mode, you owe it to yourself to hear this. Joey Lindstrom has decided
to place these in the public domain and he's asking Digium to include
them in the Asterisk prompts collection. Because he's
2013 Oct 04
1
OT: Asterisk loses Oprah on live TV
just thought this was cute enough to pass along,
https://www.youtube.com/watch?feature=player_detailpage&v=GLwct15X_3g#t=135
--
Jeremy Kister
http://jeremy.kister.net./
2010 Nov 04
2
useless mpg123 processes hanging around
Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3
when i start asterisk, i immediately see two mpg123 processes spawned
which sit there forever. I can't imagine it's normal behavior, but if
it is, please explain :)
# /etc/init.d/asterisk stop
stopping asterisk.
#[...]
# /etc/init.d/asterisk start
starting asterisk.
# psg aster
root 14573 1 0 16:29 pts/2 00:00:00
2011 Mar 29
1
wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my
ten digit "DID". I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is "asterisk".
Is this a bug? Or is there some other config I must make ?
register = 2155551941:123456 at 10.0.138.226/2155551941~600
[peer](!)
type=peer
context=inbound
qualify=yes
2009 Jan 02
1
SIP URI: Allison Smith, Music-on-Hold Parody--outstanding.
Somebody requested a path to listen without termination charges.
Here's a SIP URI: (a SIP What??)
karlonhold at sip.kfife.com or
3605195689 at 74.92.179.65
Thanks
-Karl
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2009 Feb 12
0
Friday the 13th Muhahaha Allison Smith and more on the Polycom Applications
Hi,
Allison Smith continues to contribute to the open source asterisk
resources and she is launching a new site that will make it even
easier to grab sound files. Allison joins us to talk about that and
whatever else comes up. She says:
"I have been the Voice of Asterisk -- the world's fastest-growing
telephony platform -- since its inception (which for me, was marked by
an animated
2009 Nov 13
0
VUC Today@12 ET: Allison Smith
If you missed @voicegal last time or didn't go to Astricon, join us
today on the Voip Users Conference to meet Allison Smith, the voice of
Asterisk.
Or go listen to the FBI talk about security...
http://VoipUsersConference.org for details.
/r
2009 Jan 01
5
Allison Smith, Music-on-Hold Parody--outstanding.
Allison Smith just created a hysterical parody music on hold Parody. Whatever you were doing, stop, and dial this number to listen to it: 360-519-5689. 2 minutes.
I just gave her a few ideas, but she took it and ran with it--she chose the audio and did the mix-down and everything. Really funny!!
-Karl
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2005 Jul 20
2
New voiceovers for Allison Smith: submit today
I'm sending in a set of voiceover requests to Allison Smith this
afternoon. I haven't kept up with the -users list to know if there
is someone keeping track of this stuff any more... We only have a
few phrases for her to record, and if anyone has applications which
require Allison's voice for the "asterisk-sounds" repository, let me
know. I'll be sending this in
2010 Aug 29
1
evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running
ipvoicek9-mz.124-25b.
whenever a call goes through the 1760's FXO or FXS (in or out) there is
a 915 second maximum call time due to asterisk hanging up the call
because of a "critical packet" being missed.
I read doc/sip-retransmit.txt and I don't see anything there that is
helpful to my situation - the asterisk box is
2010 Jan 10
2
app_swift 1.6.2 DTMF issue
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
for example consider:
999,1,Swift(some long message that you dont want to wait for|5000|5)
999,n,NoOp(DTMF: ${SWIFT_DTMF})
if while I am listening to the playback, i interrupt and dial:
- "12345", SWIFT_DTMF is set to
2013 Oct 10
1
asterisk 11.6 nat problem
using asterisk 11.6.0-rc1 i just converted my "nat=yes" to
"nat=auto_force_rport,auto_comedia"
I have my asterisk box on the same subnet as a cisco 1760 (vgw1).
a few times per day, Asterisk thinks vgw1 is dead (by qualify/options).
A 'sip reload' always fixes the problem.
i left 'sip set debug peer vgw1' on the console. but i dont see what's
2012 Aug 17
0
Trouble with call pickup using RPID with Cisco
I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to
asterisk 1.8.15.0.
imagining in extensions.conf:
exten => 1,1,Dial(SIP/121)
exten => 2,1,Dial(SIP/121&SIP/122)
When a caller dials extension 2 /and/ I have
trustrpid=yes
generaterpid=yes
sendrpid=yes
in sip.conf and I use the pickup exten, the caller is disconnected.
see:
2009 Dec 18
0
calls ending up in default context
I'm trying to figure out how calls are ending up in my default
context (which should never happen).
I've got a Cisco 1760V with a VIC-2FXO-M1/VIC-4FXS and 5 Cisco sip
phones.
When I make a call from one of the FXS ports on the 1760, the call
goes into asterisk's default context instead of where i think i'm
directing it.
Can someone tell me what I have misconfigured?
1760
2013 Feb 12
1
asterisk 11 AGI
I recently upgraded to asterisk 11 from 1.8.
I had VXML working via AGI in 1.8 - from extensions.conf:
[VXML]
exten => s,1,Answer
exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})})
exten => s,n,AGI(agi://localhost/url=${ENCODED})
exten => s,n,Hangup
Using asterisk 11 on the same host with the same config in extensions.conf:
-- Executing [s at VXML:1]
2015 Apr 29
0
PJSIP - sessions-timers support not working on 13.X
Ok , digging more into this i could see that (timers=no) and (timers=forced) not work asterisk not pay attention to this options when is reloaded cli not say anything and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint.
I wonder to force asterisk to refresh the session in some cases when is needed .
pjsip is able to
2020 Jun 17
0
Blog article about the state of CentOS
Il 17/06/20 09:16, Nicolas Kovacs ha scritto:
> Hi,
>
> I just read this blog article from austrian Linux expert Michael Kofler. For
> those among you who don't know the guy, he's my home country's number one Linux
> expert (known as "der Kofler") and most notably the author of a series of
> excellent books about Linux over the last 25 years.
>
>
2011 Apr 15
3
sip error logging
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk itself has been
broken.
the last entry I have is:
/var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c:
Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' -
No matching peer found
my logger.conf