Displaying 20 results from an estimated 10000 matches similar to: "g729 liscence question"
2005 Aug 25
1
where can I get low cost g723.1 liscence
Hello,
Would you please suggest me, where can I buy g723.1 liscence in cheap.
I might need a liscence for 10-50 channels.
Thanks,
2005 Aug 19
3
Sending digits from SIP to Asterisk's VoiceMailMain
Hi,
I am using Asterisk cmd VoiceMailMain to manage voice mail.
Problem is, voice mail box can't read password sent from SIP phone, but I
don't have any problem to read password from the handset attached to my
asterisk box.
Your help will be greatly appreciated.
Thanks,
2004 Dec 22
2
Out of G.729 Decoder Licenses!
Hi guys,
I got 2 licenses of g.729 and while running the asterisk with Monitor
(for recording a channel) and using one channel for the call... I
receive this error:
WARNING[23826]: codec_g729.c:180 g729tolin_framein: Out of G.729
Decoder Licenses!
many times....
it starts only when the call through the Zap channel takes place.
while this error is being running on my screen I ran the cli command:
2008 Apr 17
2
G729 license count...
I need a refresher course on how many licenses I need to buy. I have
an Asterisk server that receives calls by SIP (G729) and then sends them
to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if
the license is per channel or per call so I do not know if I need 32 or
64 licenses for this application. Could anyone please remind me?
--
Telecomunicaciones Abiertas de M?xico
2006 Dec 28
1
1.4 - G729 - Have License - No path to translate from Zap to IAX2
Hello Everybody,
Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2:
[Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729)
[Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729
[Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing
[Dec 28 21:06:00] DEBUG[1756]
2007 Oct 22
3
Authenticate by IP?
I have a customer that needs an Asterisk server to sell minutes for
cell phones in Mexico. I do not see a problem with that since he will
get the calls by SIP and then use GSM adapters to get the calls into the
GSM network. My problem is that his customers only want to be
identified by IP and not by a username and password. Is there a way to
authenticate just by using an IP address?
--
2007 Apr 20
3
pci 2.2 - pci-e x16
Hi,
Does anyone know if it is possible to plug a tdm400p pci digium card
into an pci-e 16x slot ?
Is there a possibility to work?
I have a sun fire x2100 which doesn't have pci slots.
Does Digium make pci-e cards?
2011 Jun 29
1
No audio format found to offer.
This *should* be something that's easy to fix, but apparently I'm not
doing something right.
Our SIP long distance provider is telling us to only use formats G.723
and G.729, so I've set up their trunk configuration in sip.conf as such:
[t564]
type=friend
host=XXX.XX.56.4
context=default
disallow=all
allow=g723
allow=g729
However, the Dial application gives the following error:
2005 Jul 06
11
Connect 30 phone lines to asterisk how to
Hi,
I have to connect 30 phone lines to my asterisk server, can somebody
help on how I have to do it ?
I have a TDM405P and one TDM400P with 4 FXO ports.
Do I have to use 8 TDM400P ? Or, is there another way to do it ?
Thanks,
Angel.
2008 May 20
7
Busy out a zap channel?
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would like
to busy out that line. Since that line is the head of the hunt group I
cannot simply disable that channel, I need to busy the line so calls
will come over the other lines.
--
?Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director
2005 Jul 25
3
Should this work?
Hello
I am using a Junghans quadBRI ISDN card and it is loaded and working. In Asterisk if I connect to ISDN line it is detected and tells me so.
In my zapata.conf I have (abbreviated):
[channels]
switchtype=euroisdn
signalling = bri_cpe
context=default
group=1
channel => 1-2
;plus group 2 - 4
zaptel.conf:
loadzone=uk
defaultzone=uk
# qozap span definitions
# most of the values should
2008 Feb 06
3
R2 with Alestra in Mexico...
I am trying to set up Astunicall 1.4.16 with a link from Alestra in
Mexico City. I have done everything I usually do for other links in
Mexico but this one simply will not send or receive calls. I just get
Protocol error.
Anyone has any experience with R2 and Alestra?
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
2007 May 24
3
Echo on hard SIP devices...
We have an installation with around 50 sip phones but only 5 of those are
hardware. There are three Grandstream phones, one Snom and one PAP2T. We are
running Asterisk 1.2.8 with an E1 (R2). Only the hard phones are having
problems which are either echo or distortion. The softphones all work fine
and no one is reporting any problems.
They are using 3Com switches which are fairly new. I
2012 Jun 29
3
estimating parameters of a model
Hi,
I am about to conduct an experiment of temperature dependent development of
an insect. Here I would like to know is if I could be able to find the
parameters of the models what I will use in data analysis with R program.
I have found some other softwares that can be used for parameter estimation
but liscence would be needed, not open source. So I prefer to use R langauge
if it is reliable
2005 Jun 22
3
flash panel only works with IP address
Hi,
It seems that my flash panel only works when I specify my ip address and
not the host name.
I've tried quite a few things (change host file, dns resolve,
proxying....) but couldn't get it to work.
Anyone knows how to solve this?
Thanks,
Ohad
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2005 Jun 23
5
SpanDSP - Squished Faxes
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2007 Jul 17
7
Asterisk 1.4, Unicall and Nextel...
I have a customer that is complaining that any call coming in from
Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel
1.4.3 and all the MFC/R2 patches and libraries. All other calls go out
and come in, just Nextel seems to have this problem. The phone company
technician connected a PBX emulator on the line and that one could
receive the calls from Nextel.
The E1 is provided
2006 Dec 20
5
Sangoma A101 with Unicall
I am having a problem trying to get a Sangoma A101 to work with
Unicall. I have installed the sangoma drivers and everything seems to
be well but when I try to run ztcfg I get the following error:
CAS signalling on span 1 conflicts with HDLC with FCS check on channel
16.
Here is my /etc/zaptel.conf
# MFC/R2 normalmente no usa CRC4
span=1,0,0,cas,hdb3
cas=1-15:1101
dchan=16
cas=17-31:1101
2006 Dec 06
2
problem with asterisk - calls where both sidescannot hear each other
If you use both the public and private interfaces for VoIP in the Asterisk Server, make sure you don't specify one of them for the binding in sip.conf
Example
bindaddr=0.0.0.0
will allow SIP traffic on any of your interfaces.
Ed Nu?ez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-----Original Message-----
From:
2003 Oct 03
1
Budgettone + G729
hi there ..
I asked sometime ago regarding getting a Budgettone
working with Asterisk over G729.
My system is quite simple, Asterisk server with 1 G 729 license
installed, and 10 Grandstream phones. Only one of them needs
G729, because it's on a remote link via an ADSL bridge. The
rest run happily on G711 on a local network.
I added the lines
disallow=all
allow=g729
to the sip.conf entry