Displaying 20 results from an estimated 20 matches for "asteriskforum".
2008 Jan 16
1
Asterisk 1.4.17 and RXFAX via T38
I was pointed to the following:
http://asteriskforum.ru/viewtopic.php?t=1761
It is in Russian, which I don't speak, but it references an Asterisk patch.
Is this patch in 1.4.17?
Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?)
Anyone work with this?
2009 Sep 18
3
DUNDi + SIP Realtime
...9;m doing terribly wrong, that would break everything and make the universe collapse into itself when I apply the same principle on production?
I'll be happy to provide more details in case there are any doubts. I really appreciate your feedback, no matter what is it. :)
Vin?cius Fontes
www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP
2009 Jan 14
1
1.6.1-b4: Can't get fax2mail work from System()
...n/fax2mail.1.sh --dest-name "Sean" --dest-email
"seandarcy2 at gmail.com" -f "/var/spool/asterisk/fax/200901141711-0"
Am I screwing up the System() command somehow? Is System() screwed up in
1.6.1?
Any clues how to debug this? I did find one relevant thread
http://asteriskforum.ru/viewtopic.php?p=15629 , which is unfortunatley
in Russian. In that thread someone figured out how to turn on DEBUG for
app_fax. How did you do that?
sean
2009 Sep 02
4
More Echo
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<span class="postbody">Greetings,<br>
I am running Asterisk 1.4.25 with Dahdi Complete
2.2.0, on a Digium TE121B PCI express card with a </span>VPMADT032<span
2009 Sep 29
1
Fax and dial-up connection issues
...326632 9071480 Rescheduling interrupts
CAL: 303 529 function call interrupts
TLB: 1521462 4810165 TLB shootdowns
TRM: 0 0 Thermal event interrupts
SPU: 0 0 Spurious interrupts
ERR: 0
MIS: 0
Vin?cius Fontes
www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP
2009 Mar 02
3
Dialing with cli
Any way to initiate a call and execute a playback of an audio file from the cli?
My only chance to debug or make changes is usually when no one's at the office including me!
Thanks!
jlc
2009 Jul 22
1
grandstream and jitter buffer
Hi guys,
I have a bunch grandstream phones using ulaw and my users are
complaining they are jittery when I use "canreinvite=yes". The data
connection is an ADSL link dedicated for phone traffic. At any given
time, I have at most 2 calls in parallel.
I'm not a huge fan of asterisk being in media path doing buffering
because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and
2009 Nov 06
1
Need opinion about GSM codec for Internet
Dear all, I have implemented an Asterisk SIP server for a WAN VPN over
Internet. We have users distributed along all my country (Argentina) that
register to my Asterisk in order to talk among them.
I'll plan to have voice and voicemail with GSM codec, because we can't
afford the payment for the G.729 licenses (it's an administrative problem of
our company, not an echonomical problem).
2009 Dec 14
1
Call on hold through DTMF
Hi everybody,
I have a sip phone (Siemens) which has no sip functions at all.
Is is possible to press #4 by example to put the call on hold then
dial #2 to get the call back ?
I'have look at features.conf but i did not find the solution.
I know the call parking functionnality, but i would like a much simple
system.
I hope i'm clear enough.
Thank you
Matthieu NICAISE
Responsable
2009 Sep 14
3
G.729 for Asterisk
Hello
I have a confusion relating to G.729 codec.
I know how to install where to get license but i really don't know why we
need it?
Why people use G.729 codec with asterisk?
look all functionality can be done with out it ie calling from sip to iax
protocol and sip/ iax to E1, then why we need this??
regards
Adam
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2009 Oct 14
2
DAHDI Dummy for Linux VServers
I'm running dahdi on the host system, and have added the /dev/dahdi/
devices to the guest vserver as recommended in Beave's "Virtual Private
Asterisk" whitepaper (http://www.telephreak.org/papers/vpa/).
I tried copying libtonezone.so and libtonezone.h to the guest, but I
couldn't anything to replace zaptel.h in DAHDI souce (it seems dahdi.h
was deprecated?).
I need to
2009 Dec 29
1
T.38 and Linksys SPA8000
...tpsource=yes
[9204]
type=friend
secret=******
callerid="" <9204>
host=dynamic
nat=no
canreinvite=yes
insecure=port,invite
qualify=yes
context=ddi
callgroup=1
pickupgroup=1
call-limit=10
disallow=all
allow=g729
allow=alaw
t38pt_udptl=yes
t38pt_usertpsource=yes
Vin?cius Fontes
www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP
2009 May 21
2
Jitter buffer question
Hi List,
I have a question regarding jitterbuffer in Asterisk 1.4.24. I see that
jitterbuffer is only effective on the receiving channels.
My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch
office.
Questions:
1. To enable jitter buffer on SIP channels it seems I have to enable and
force it, right?
2. If I enable and force jitter buffer, Asterisk would always have to
stay
2009 Dec 14
3
Question regarding digital card TE412p
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am planning to use TE412p (includes echo cancellation) 4 port digital card
(PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI
connections) with proper hardware like dual core quadcore processor and 8gb
RAM in one server?
Also I was planning to implement using 64 bit architecture with Asterisk:
2009 Sep 22
5
New Xorcom FXS USB Bank is not loading firmware
Hi,
I just got a Xorcom Astribank with 8 FXS but it does not work. So I
tried resetting and loading the firmware. But loading just times out.
/usr/share/zaptel/xpp_fxloader usb
--------- FIRMWARE LOADING: (usb) [1 devices]
'xpp_fxloader'[11561]: USB Firmware /usr/share/zaptel/USB_FW.hex
into /dev/bus/usb/002/003
2009 Dec 14
3
Is this bad hardware? Dahdi-v-X100 clone
I've spent a week playing with Asterisk 1.6 and I love it. What a
brilliant piece of software!
Progress and learning have been reasonably good. I have external SIP
provider calls coming in and have put together a little call platform
and I'm stunned at the flexibility.
There is one issue for me. I took me a while to click that ZAPTEL now
equals Dahdi, but now I'm there I have an
2009 Sep 23
3
SIP/WiFi handsets?
Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent
price:quality ratio, of possible. Keep seeing handsets for Vonage, etc.,
in Best Buy and the like, but I imagine it's locked to Vonage, and can't
be re-appropriated.
Thanks!
-Ken
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2009 Nov 15
6
Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server
Dear All
Please be informed that I need to install Asterisk 1.4.13 on my Debian 3.1
server . But I got the following message when trying for "#./configure" :
"error: no acceptable C compiler found in $PATH"
Can you please do me favor and let me know what is the problem ?
Let me thank you in advance
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2009 Jun 04
7
Asterisk AGI issues (at high load)
Hi, we are experiencing a strange issue and I am hoping someone can point me
to the right direction or help out with some pointers.
We have asterisk 1.6.0.6 with a sangome a104DE card. We have basically 4
T1's for a total of DAHDI 96 channels.
We have an agi application (php) that acts as a kind of a calling card
application.
All users are SIP users that make calls and asterisk then bridges
2008 Jan 17
0
Channels ID / Soft Hang Up
...nd RXFAX via T38
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478E67E4.8070405 at htt-consult.com>
Content-Type: text/plain; charset="US-ASCII"; format="flowed"
I was pointed to the following:
http://asteriskforum.ru/viewtopic.php?t=1761
It is in Russian, which I don't speak, but it references an Asterisk
patch.
Is this patch in 1.4.17?
Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?)
Anyone work with this?
------------------------------
Message: 8
Date: Wed, 16 Jan 2008 20:39:5...