Displaying 20 results from an estimated 10000 matches similar to: "Dialing with cli"
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 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
2008 May 05
3
TDM410P driver?
Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver?
Att
Vin?cius Fontes
Desenvolvimento
Canall Tecnologia em Comunica??es Ltda.
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 29
1
Fax and dial-up connection issues
I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones.
The problem is that fax and dial-up connections are really
2009 Sep 18
3
DUNDi + SIP Realtime
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729
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 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:
2008 Oct 25
9
Cheapest 4 port FXO
I need to increase reliability at an office as SIP/Internet provider outages are causing some issues.
What would be the least expensive analogue card that people are using reliably?
Thanks!
jlc
2010 Jul 12
6
Project Management Solutions
I need to implement a solution and not having ever used anything but
MS Project I would be grateful for a reco on something good. The only
hope would be that its web based but I am open to anything!
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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2010 Feb 05
6
Still on spandsp/app_fax and T.38
This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :)
Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it
2008 Apr 07
7
MS Exchange Replacement
What is the closest open source mail server I can replace exchange with that provides the nearest equivalent in user experience?
Thanks!
jlc
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2010 Feb 27
5
Python Script Issue
Hey Guys,
I am trying to get a python script running but I get the following error:
atexit.register(atexit_handler)
NameError: global name 'atexit' is not defined
A Google search doesn't really help me, except confirm that I don't know
anything about python:)
Anyone got any ideas?
Thanks,
jlc
2008 Apr 24
1
Full queue issues
Hello everyone.
I got a little problem in here: I want to set up a queue so that if anything of these happens:
a) No agents logged in
b) All agents busy
Then the user gets diverted somewhere. I used this (for testing purposes only, of course):
exten => 7080,1,Answer()
exten => 7080,n,Queue(teste)
exten => 7080,n,Goto(${QUEUESTATUS})
exten => 7080,n(ERROR),NoOp(${QUEUESTATUS})
exten
2009 Feb 27
16
(a) WinPower RPM available? (b) where to install if use tar.gz file?
Two quick and simple questions; I want to install the "WinPower"
software for my new UPS. On their web site, they have a tar.gz file
available for download. I know the reasons for staying with RPM, if at
all possible. I''ve Googled and Yahood for ""WinPower"+RPM+Linux and
get hits, but no obvious RPM. I tried "yum install winpower" and the
response was
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